Internet Engineering Task Force SIP WG Internet Draft Jonathan Rosenberg dynamicsoft Henning Schulzrinne Columbia U. Gonzalo Camarillo Ericsson Alan Johnston Worldcom Jon Peterson Neustar Robert Sparks dynamicsoft Mark Handley ACIRI Eve Schooler AT&T draft-ietf-sip-rfc2543bis-06.txt January 28, 2002 Expires: July 2002 SIP: Session Initiation Protocol STATUS OF THIS MEMO This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress". The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt To view the list Internet-Draft Shadow Directories, see http://www.ietf.org/shadow.html. Abstract The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution and multimedia conferences. SIP invitations used to create sessions carry session descriptions which allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the users current location, authenticate and authorize users for services, implement provider call routing policies, and provide features to users. SIP also provides a registration function that allows them to upload their current location for use by proxy servers. SIP runs ontop of several different transport protocols. Various Authors [Page a] Internet Draft SIP January 28, 2002 Table of Contents 1 Introduction ........................................ 2 2 Overview of SIP Functionality ....................... 2 3 Terminology ......................................... 3 4 Overview of Operation ............................... 4 5 Structure of the Protocol ........................... 11 6 Definitions ......................................... 13 7 SIP Messages ........................................ 19 7.1 Requests ............................................ 20 7.2 Responses ........................................... 20 7.3 Header Fields ....................................... 21 7.3.1 Header Field Format ................................. 22 7.3.2 Header Field Classification ......................... 24 7.3.3 Compact Form ........................................ 25 7.4 Bodies .............................................. 25 7.4.1 Message Body Type ................................... 25 7.4.2 Message Body Length ................................. 25 7.5 Framing SIP messages ................................ 26 8 General User Agent Behavior ......................... 26 8.1 UAC Behavior ........................................ 27 8.1.1 Generating the Request .............................. 27 8.1.1.1 Request-URI ......................................... 27 8.1.1.2 To .................................................. 27 8.1.1.3 From ................................................ 28 8.1.1.4 Call-ID ............................................. 29 8.1.1.5 CSeq ................................................ 30 8.1.1.6 Max-Forwards ........................................ 30 8.1.1.7 Via ................................................. 31 8.1.1.8 Contact ............................................. 31 8.1.1.9 Supported and Require ............................... 32 8.1.1.10 Additional Message Components ....................... 32 8.1.2 Sending the Request ................................. 33 8.1.3 Loose Routing Policies .............................. 33 8.1.3.1 Modifying the Route header field .................... 33 8.1.3.2 Modifying the Request-URI ........................... 34 8.1.3.3 Destination Choice .................................. 34 8.1.3.4 Loop Avoidance ...................................... 34 8.1.4 Processing Responses ................................ 35 8.1.4.1 Transaction Layer Errors ............................ 35 8.1.4.2 Unrecognized Responses .............................. 35 8.1.4.3 Vias ................................................ 36 8.1.4.4 Processing Reliable 1xx Responses ................... 36 Various Authors [Page b] Internet Draft SIP January 28, 2002 8.1.4.5 Processing 3xx responses ............................ 36 8.1.4.6 Processing 4xx responses ............................ 38 8.2 UAS Behavior ........................................ 39 8.2.1 Method Inspection ................................... 39 8.2.2 Header Inspection ................................... 39 8.2.2.1 To and Request-URI .................................. 39 8.2.2.2 Merged Requests ..................................... 40 8.2.2.3 Require ............................................. 40 8.2.3 Content Processing .................................. 41 8.2.4 Applying Extensions ................................. 42 8.2.5 Processing the Request .............................. 42 8.2.6 Generating the Response ............................. 42 8.2.6.1 Sending a Provisional Response ...................... 42 8.2.6.2 Headers and Tags .................................... 43 8.2.7 Stateless UAS Behavior .............................. 43 8.3 Redirect Servers .................................... 44 9 Canceling a Request ................................. 45 9.1 Client Behavior ..................................... 46 9.2 Server Behavior ..................................... 47 10 Registrations ....................................... 48 10.1 Overview ............................................ 48 10.2 Constructing the REGISTER Request ................... 49 10.2.1 Adding Bindings ..................................... 52 10.2.1.1 Setting the Expiration Interval of Contact Addresses ...................................................... 52 10.2.1.2 Preferences among Contact Addresses ................. 53 10.2.2 Removing Bindings ................................... 53 10.2.3 Fetching Bindings ................................... 53 10.2.4 Refreshing Bindings ................................. 53 10.2.5 Setting the Internal Clock .......................... 54 10.2.6 Discovering a Registrar ............................. 54 10.2.7 Transmitting a Request .............................. 55 10.2.8 Error Responses ..................................... 55 10.3 Processing REGISTER Requests ........................ 55 11 Querying for Capabilities ........................... 58 11.1 Construction of OPTIONS Request ..................... 59 11.2 Processing of OPTIONS Request ....................... 59 12 Dialogs ............................................. 61 12.1 Creation of a Dialog ................................ 62 12.1.1 UAS behavior ........................................ 62 12.1.2 UAC behavior ........................................ 63 12.2 Requests within a Dialog ............................ 64 12.2.1 UAC Behavior ........................................ 65 12.2.1.1 Generating the Request .............................. 65 12.2.1.2 Processing the Responses ............................ 66 12.2.2 UAS behavior ........................................ 67 12.3 Termination of a Dialog ............................. 69 13 Initiating a Session ................................ 69 Various Authors [Page c] Internet Draft SIP January 28, 2002 13.1 Overview ............................................ 69 13.2 Caller Processing ................................... 70 13.2.1 Creating the Initial INVITE ......................... 70 13.2.2 Processing INVITE Responses ......................... 72 13.2.2.1 1xx responses ....................................... 72 13.2.2.2 3xx responses ....................................... 72 13.2.2.3 4xx, 5xx and 6xx responses .......................... 72 13.2.2.4 2xx responses ....................................... 73 13.3 Callee Processing ................................... 74 13.3.1 Processing of the INVITE ............................ 74 13.3.1.1 Progress ............................................ 75 13.3.1.2 The INVITE is redirected ............................ 75 13.3.1.3 The INVITE is rejected .............................. 76 13.3.1.4 The INVITE is accepted .............................. 76 14 Modifying an Existing Session ....................... 77 14.1 UAC Behavior ........................................ 77 14.2 UAS Behavior ........................................ 79 15 Terminating a Session ............................... 80 15.1 Terminating a Dialog with a BYE Request ............. 81 15.1.1 UAC Behavior ........................................ 81 15.1.2 UAS Behavior ........................................ 82 16 Proxy Behavior ...................................... 82 16.1 Overview ............................................ 82 16.2 Stateful Proxy ...................................... 83 16.3 Request Validation .................................. 84 16.4 Making a Routing Decision ........................... 87 16.5 Request Processing .................................. 90 16.6 Response Processing ................................. 97 16.7 Processing Timer C .................................. 105 16.8 Handling Transport Errors ........................... 105 16.9 CANCEL Processing ................................... 105 16.10 Stateless Proxy ..................................... 106 16.11 Record-Route Example ................................ 108 17 Transactions ........................................ 109 17.1 Client Transaction .................................. 111 17.1.1 INVITE Client Transaction ........................... 112 17.1.1.1 Overview of INVITE Transaction ...................... 112 17.1.1.2 Formal Description .................................. 113 17.1.1.3 Construction of the ACK Request ..................... 116 17.1.2 non-INVITE Client Transaction ....................... 117 17.1.2.1 Overview of the non-INVITE Transaction .............. 117 17.1.2.2 Formal Description .................................. 117 17.1.3 Matching Responses to Client Transactions ........... 118 17.1.4 Handling Transport Errors ........................... 120 17.2 Server Transaction .................................. 120 17.2.1 INVITE Server Transaction ........................... 120 17.2.2 non-INVITE Server Transaction ....................... 123 17.2.3 Matching Requests to Server Transactions ............ 124 Various Authors [Page d] Internet Draft SIP January 28, 2002 17.2.4 Handling Transport Errors ........................... 126 17.3 RTT Estimation ...................................... 126 18 Reliability of Provisional Responses ................ 127 18.1 UAS Behavior ........................................ 128 18.2 UAC Behavior ........................................ 130 19 Transport ........................................... 131 19.1 Clients ............................................. 132 19.1.1 Sending Requests .................................... 132 19.1.2 Receiving Responses ................................. 134 19.2 Servers ............................................. 134 19.2.1 Receiving Requests .................................. 134 19.2.2 Sending Responses ................................... 135 19.3 Framing ............................................. 136 19.4 Error Handling ...................................... 136 20 Usage of HTTP Authentication ........................ 137 20.1 Framework ........................................... 137 20.2 User-to-User Authentication ......................... 139 20.3 Proxy to User Authentication ........................ 141 20.4 The Digest Authentication Scheme .................... 143 20.4.1 HTTP Digest ......................................... 143 21 S/MIME .............................................. 145 21.1 S/MIME Certificates ................................. 145 21.2 S/MIME Key Exchange ................................. 146 21.3 Securing MIME bodies ................................ 148 21.4 Tunneling SIP in MIME ............................... 149 21.4.1 Tunneling Integrity and Authentication .............. 149 21.4.2 Tunneling Encryption ................................ 151 22 Security Considerations ............................. 152 22.1 Threat Models ....................................... 153 22.1.1 Registration Hijacking .............................. 153 22.1.2 Impersonating a Server .............................. 154 22.1.3 Tampering with Message Bodies ....................... 154 22.1.4 Tearing Down Sessions ............................... 155 22.1.5 Denial of Service and Amplification ................. 156 22.2 Security Mechanisms ................................. 156 22.2.1 Transport and Network Layer Security ................ 157 22.2.2 HTTP Authentication ................................. 158 22.2.3 S/MIME .............................................. 158 22.3 Implementing Security Mechanisms .................... 159 22.3.1 Requirements for Implementers of SIP ................ 159 22.3.2 Security Solutions .................................. 160 22.3.2.1 Registration ........................................ 160 22.3.2.2 Requests and Transitive Trust ....................... 161 22.3.2.3 Peer to Peer Requests ............................... 163 22.3.2.4 DoS Protection ...................................... 164 22.4 Limitations ......................................... 165 22.4.1 HTTP Digest ......................................... 165 22.4.2 S/MIME .............................................. 166 Various Authors [Page e] Internet Draft SIP January 28, 2002 22.4.3 TLS ................................................. 167 22.5 Privacy ............................................. 167 23 Common Message Components ........................... 168 23.1 SIP Uniform Resource Indicators ..................... 168 23.1.1 SIP URI Components .................................. 168 23.1.2 Character Escaping Requirements ..................... 172 23.1.3 Example SIP URIs .................................... 172 23.1.4 SIP URI Comparison .................................. 173 23.1.5 Forming Requests from a SIP URI ..................... 175 23.1.6 Relating SIP URIs and tel URLs ...................... 176 23.2 Option Tags ......................................... 178 23.3 Tags ................................................ 179 24 Header Fields ....................................... 179 24.1 Accept .............................................. 181 24.2 Accept-Encoding ..................................... 181 24.3 Accept-Language ..................................... 184 24.4 Alert-Info .......................................... 184 24.5 Allow ............................................... 184 24.6 Authentication-Info ................................. 185 24.7 Authorization ....................................... 185 24.8 Call-ID ............................................. 186 24.9 Call-Info ........................................... 186 24.10 Contact ............................................. 186 24.11 Content-Disposition ................................. 187 24.12 Content-Encoding .................................... 188 24.13 Content-Language .................................... 189 24.14 Content-Length ...................................... 189 24.15 Content-Type ........................................ 189 24.16 CSeq ................................................ 190 24.17 Date ................................................ 190 24.18 Error-Info .......................................... 191 24.19 Expires ............................................. 191 24.20 From ................................................ 191 24.21 In-Reply-To ......................................... 192 24.22 Max-Forwards ........................................ 192 24.23 Min-Expires ......................................... 193 24.24 MIME-Version ........................................ 193 24.25 Organization ........................................ 193 24.26 Priority ............................................ 194 24.27 Proxy-Authenticate .................................. 194 24.28 Proxy-Authorization ................................. 195 24.29 Proxy-Require ....................................... 195 24.30 RAck ................................................ 195 24.31 Record-Route ........................................ 196 24.32 Reply-To ............................................ 196 24.33 Require ............................................. 196 24.34 Retry-After ......................................... 197 24.35 Route ............................................... 197 Various Authors [Page f] Internet Draft SIP January 28, 2002 24.36 RSeq ................................................ 198 24.37 Server .............................................. 198 24.38 Subject ............................................. 198 24.39 Supported ........................................... 199 24.40 Timestamp ........................................... 199 24.41 To .................................................. 199 24.42 Unsupported ......................................... 200 24.43 User-Agent .......................................... 200 24.44 Via ................................................. 200 24.45 Warning ............................................. 201 24.46 WWW-Authenticate .................................... 203 25 Response Codes ...................................... 203 25.1 Provisional 1xx ..................................... 204 25.1.1 100 Trying .......................................... 204 25.1.2 180 Ringing ......................................... 204 25.1.3 181 Call Is Being Forwarded ......................... 204 25.1.4 182 Queued .......................................... 204 25.1.5 183 Session Progress ................................ 204 25.2 Successful 2xx ...................................... 205 25.2.1 200 OK .............................................. 205 25.3 Redirection 3xx ..................................... 205 25.3.1 300 Multiple Choices ................................ 205 25.3.2 301 Moved Permanently ............................... 205 25.3.3 302 Moved Temporarily ............................... 206 25.3.4 305 Use Proxy ....................................... 206 25.3.5 380 Alternative Service ............................. 206 25.4 Request Failure 4xx ................................. 206 25.4.1 400 Bad Request ..................................... 206 25.4.2 401 Unauthorized .................................... 207 25.4.3 402 Payment Required ................................ 207 25.4.4 403 Forbidden ....................................... 207 25.4.5 404 Not Found ....................................... 207 25.4.6 405 Method Not Allowed .............................. 207 25.4.7 406 Not Acceptable .................................. 207 25.4.8 407 Proxy Authentication Required ................... 207 25.4.9 408 Request Timeout ................................. 208 25.4.10 410 Gone ............................................ 208 25.4.11 413 Request Entity Too Large ........................ 208 25.4.12 414 Request-URI Too Long ............................ 208 25.4.13 415 Unsupported Media Type .......................... 208 25.4.14 416 Unsupported URI Scheme .......................... 208 25.4.15 420 Bad Extension ................................... 208 25.4.16 421 Extension Required .............................. 209 25.4.17 423 Registration Too Brief .......................... 209 25.4.18 480 Temporarily Unavailable ......................... 209 25.4.19 481 Call/Transaction Does Not Exist ................. 209 25.4.20 482 Loop Detected ................................... 210 25.4.21 483 Too Many Hops ................................... 210 Various Authors [Page g] Internet Draft SIP January 28, 2002 25.4.22 484 Address Incomplete .............................. 210 25.4.23 485 Ambiguous ....................................... 210 25.4.24 486 Busy Here ....................................... 211 25.4.25 487 Request Terminated .............................. 211 25.4.26 488 Not Acceptable Here ............................. 211 25.4.27 491 Request Pending ................................. 211 25.4.28 493 Undecipherable .................................. 211 25.5 Server Failure 5xx .................................. 211 25.5.1 500 Server Internal Error ........................... 211 25.5.2 501 Not Implemented ................................. 212 25.5.3 502 Bad Gateway ..................................... 212 25.5.4 503 Service Unavailable ............................. 212 25.5.5 504 Server Time-out ................................. 212 25.5.6 505 Version Not Supported ........................... 212 25.5.7 513 Message Too Large ............................... 213 25.6 Global Failures 6xx ................................. 213 25.6.1 600 Busy Everywhere ................................. 213 25.6.2 603 Decline ......................................... 213 25.6.3 604 Does Not Exist Anywhere ......................... 213 25.6.4 606 Not Acceptable .................................. 213 26 Examples ............................................ 214 26.1 Registration ........................................ 214 26.2 Session Setup ....................................... 215 27 Augmented BNF for the SIP Protocol ................. 220 27.1 Basic Rules ......................................... 222 28 IANA Considerations ............................ 239 28.1 Option Tags ......................................... 239 28.1.1 Registration of 100rel .............................. 240 28.2 Warn-Codes .......................................... 241 28.3 Header Field Names .................................. 241 28.4 Method and Response Codes ........................... 242 29 Changes Made in Version 00 .......................... 242 30 Changes Made in Version 01 .......................... 249 31 Changes Made in Version 02 .......................... 249 32 Changes Made in Version 03 .......................... 251 33 Changes Made in Version 04 .......................... 254 34 Changes Made in Version 05 .......................... 256 35 Changes Made in Version 06 .......................... 260 36 Acknowledgments ..................................... 272 37 Authors' Addresses .................................. 272 38 Bibliography ........................................ 274 EOTOC Various Authors [Page h] 1 Introduction There are many applications of the Internet that require the creation and management of a session, where a session is considered an exchange of data between an association of participants. The implementation of these services is complicated by the practices of participants; users may move between endpoints, they may be addressable by multiple names, and they may communicate in several different media - sometimes simultaneously. Numerous protocols have been authored that carry various forms of real-time multimedia session data such as voice, video, or text messages. SIP works in concert with these protocols by enabling Internet endpoints (called "user agents") to discover one another and to agree on a characterization of a session they would like to share. For locating prospective session participants, and for other functions, SIP enables creation of an infrastructure of network hosts (called "proxy servers") to which user agents can send registrations, invitations to sessions and other requests. SIP is an agile, general-purpose tool for creating, modifying and terminating sessions that works independently of underlying transport protocols and without dependency on the type of session that is being established. 2 Overview of SIP Functionality The Session Initiation Protocol (SIP) is an application-layer control protocol that can establish, modify, and terminate multimedia sessions (conferences) such as Internet telephony calls. SIP can also invite participants to already existing sessions, such as multicast conferences. Media can be added to (and removed from) an existing session. SIP transparently supports name mapping and redirection services, which supports personal mobility [1] - users can maintain a single externally visible identifier (SIP URI) regardless of their network location. SIP supports five facets of establishing and terminating multimedia communications: User location: determination of the end system to be used for communication; User availability: determination of the willingness of the called party to engage in communications; User capabilities: determination of the media and media parameters to be used; Session setup: "ringing", establishment of session parameters at both called and calling party; Various Authors [Page 2] Internet Draft SIP January 28, 2002 Session management: including transfer and termination of sessions, modifying session parameters, and invoking services. SIP is not a vertically integrated communications system. SIP is rather a component that can be used with other IETF protocols to build a complete multimedia architecture. Typically, these architectures will include protocols such as the real-time transport protocol (RTP) (RFC 1889 [2]) for transporting real-time data and providing QoS feedback, the real-time streaming protocol (RTSP) (RFC 2326 [3]) for controlling delivery of streaming media, the Media Gateway Control Protocol (MEGACO) (RFC 3015 [4]) for controlling gateways to the Public Switched Telephone Network (PSTN), and the session description protocol (SDP) (RFC 2327 [5]) for describing multimedia sessions. Therefore, SIP should be used in conjunction with other protocols in order to provide complete services to the users. However, the basic functionality and operation of SIP does not depend on any of these protocols. SIP does not provide services. SIP rather provides primitives that can be used to implement different services. For example, SIP can locate a user and deliver an opaque object to his current location. If this primitive is used to deliver a session description written in SDP, for instance, the parameters of a session can be agreed between endpoints. If the same primitive is used to deliver a photo of the caller as well as the session description, a "caller ID" service can be easily implemented. As this example shows, a single primitive is typically used to provide several different services. SIP does not offer conference control services such as floor control or voting and does not prescribe how a conference is to be managed. SIP can be used to initiate a session that uses some other conference control protocol. Since SIP messages and the sessions they establish can pass through entirely different networks, SIP cannot, and does not, provide any kind of network resource reservation capabilities. The nature of the services provided by SIP make security particularly important. To that end, SIP provides a suite of security services, which include denial-of-service prevention, authentication (both user to user and proxy to user), integrity protection, and encryption and privacy services. SIP works with both IPv4 and IPv6. 3 Terminology In this document, the key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALLNOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", Various Authors [Page 3] Internet Draft SIP January 28, 2002 and "OPTIONAL" are to be interpreted as described in RFC 2119 [6] and indicate requirement levels for compliant SIP implementations. 4 Overview of Operation This section introduces the basic operations of SIP using simple examples. This section is tutorial in nature and does not contain any normative statements. The first example shows the basic functions of SIP: location of an end point, signal of a desire to communicate, negotiation of session parameters to establish the session, and teardown of the session once established. Figure 1 shows a typical example of a SIP message exchange between two users, Alice and Bob. (Each message is labeled with the letter "F" and a number for reference by the text.) In this example, Alice uses a SIP application on her PC (referred to as a softphone) to call Bob on his SIP phone over the Internet. Also shown are two SIP proxy servers that act on behalf of Alice and Bob to facilitate the session establishment. This typical arrangement is often referred to as the "SIP trapezoid" as shown by the geometric shape of the dashed lines in Figure 1. Alice "calls" Bob using his SIP identity, a type of Uniform Resource Identifier (URI) called a SIP URI and defined in Section 23.1. It has a similar form to an email address, typically containing a username and a host name. In this case, it is sip:bob@biloxi.com, where biloxi.com is the domain of Bob's SIP service provider (which can be an enterprise, retail provider, etc). Alice also has a SIP URI of sip:alice@atlanta.com. Alice might have typed in Bob's URI or perhaps clicked on a hyperlink or an entry in an address book. SIP is based on an HTTP-like request/response transacton model. Each transaction consists of a request that invokes a particular "Method", or function, on the server, and at least one response. In this example, the transaction begins with Alice's softphone sending an INVITE request addressed to Bob's SIP URI. INVITE is an example of a SIP method which specifies the action that the requestor (Alice) wants the server (Bob) to take. The INVITE request contains a number of header fields. Header fields are named attributes that provide additional information about a message. The ones present in an INVITE include a unique identifier for the call, the destination address, Alice's address, and information about the type of session that Alice wishes to establish with Bob. The INVITE (message F1 in Figure 1) might look like this: Various Authors [Page 4] Internet Draft SIP January 28, 2002 atlanta.com . . . biloxi.com . proxy proxy . . . Alice's . . . . . . . . . . . . . . . . . . . . Bob's softphone SIP Phone | | | | | INVITE F1 | | | |--------------->| INVITE F2 | | | 100 Trying F3 |--------------->| INVITE F4 | |<---------------| 100 Trying F5 |--------------->| | |<-------------- | 180 Ringing F6 | | | 180 Ringing F7 |<---------------| | 180 Ringing F8 |<---------------| 200 OK F9 | |<---------------| 200 OK F10 |<---------------| | 200 OK F11 |<---------------| | |<---------------| | | | ACK F12 | |------------------------------------------------->| | Media Session | |<================================================>| | BYE F13 | |<-------------------------------------------------| | 200 OK F14 | |------------------------------------------------->| | | Figure 1: SIP session setup example with SIP trapezoid INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds To: Bob From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: Content-Type: application/sdp Content-Length: 142 (Alice's SDP not shown) Various Authors [Page 5] Internet Draft SIP January 28, 2002 The first line of the text-encoded message contains the method name (INVITE). The lines that follow are a list of header fields. This example contains a minimum required set. The headers are briefly described below: Via contains the address (pc33.atlanta.com) on which Alice is expecting to receive responses to this request.It also contains a branch parameter that contains an identifier for this transaction. To contains a display name (Bob) and a SIP URI (sip:bob@biloxi.com) towards which the request was originally directed. Display names are described in RFC 2822 [7]. From also contains a display name (Alice) and a SIP URI (sip:alice@atlanta.com) that indicate the originator of the request. This header field also has a tag parameter containing a pseudorandom string (1928301774) that was added to the URI by the softphone. It is used for identification purposes. Call-ID contains a globally unique identifier for this call, generated by the combination of a pseudorandom string and the softphone's IP address. The combination of the To, From, and Call-ID completely define a peer-to-peer SIP relationship betwee Alice and Bob, and is referred to as a "dialog". CSeq or Command Sequence contains an integer and a method name. The CSeq number is incremented for each new request, and is a traditional sequence number. Contact contains a SIP URI that represents a direct route to reach or contact Alice, usually composed of a username at an FQDN. While a FQDN is preferred, many end systems do not have registered domain names, so IP addresses are permitted. While the Via header field tells other elements where to send the response, the Contact header field tells other elements where to send future requests for this dialog. Content-Type contains a description of the message body (not shown). Content-Length contains an octet (byte) count of the message body. The complete set of SIP header fields is defined in Section 24. The details of the session, type of media, codec, sampling rate, etc. are not described using SIP. Rather, the body of a SIP message contains a description of the session, encoded in some other protocol format. One such format is Session Description Protocol (SDP) [5]. This SDP message (not shown in the example) is carried by the SIP Various Authors [Page 6] Internet Draft SIP January 28, 2002 message in a way that is analogous to a document attachment being carried by an email message, or a web page being carried in an HTTP message. Since the softphone does not know the location of Bob or the SIP server in the biloxi.com domain, the softphone sends the INVITE to the SIP server that serves Alice's domain, atlanta.com. The IP address of the atlanta.com SIP server could have been configured in Alice's softphone, or it could have been discovered by DHCP, for example. The atlanta.com SIP server is a type of SIP server known as a proxy server. A proxy server receives SIP requests and forwards them on behalf of the requestor. In this example, the proxy server receives the INVITE request and sends a 100 (Trying) response back to Alice's softphone. The 100 (Trying) response indicates that the INVITE has been received and that the proxy is working on her behalf to route the INVITE to the destination. Responses in SIP use a three-digit code followed by a descriptive phrase. This response contains the same To, From, Call-ID, and CSeq as the INVITE, which allows Alice's softphone to correlate this response to the sent INVITE. The atlanta.com proxy server locates the proxy server at biloxi.com, possibly by performing a particular type of DNS (Domain Name Service) lookup to find the SIP server that serves the biloxi.com domain. This is described in [8]. As a result, it obtains the IP address of the biloxi.com proxy server and forwards, or proxies, the INVITE request there. Before forwarding the request, the atlanta.com proxy server adds an additional Via header field that contains its own IP address (the INVITE already contains Alice's IP address in the first Via). The biloxi.com proxy server receives the INVITE and responds with a 100 (Trying) response back to the Atlanta.com proxy server to indicate that it has received the INVITE and is processing the request. The proxy server consults a database, generically called a location service, that contains the current IP address of Bob. (We shall see in the next section how this database can be populated.) The biloxi.com proxy server adds another Via header with its own IP address to the INVITE and proxies it to Bob's SIP phone. Bob's SIP phone receives the INVITE and alerts Bob to the incoming call from Alice so that Bob can decide whether or not to answer the call, i.e., Bob's phone rings. Bob's SIP phone sends an indication of this in a 180 (Ringing) response, which is routed back through the two proxies in the reverse direction. Each proxy uses the Via header to determine where to send the response and removes its own address from the top. As a result, although DNS and location service lookups were required to route the initial INVITE, the 180 (Ringing) response can be returned to the caller without lookups or without state being maintained in the proxies. This also has the desirable property that Various Authors [Page 7] Internet Draft SIP January 28, 2002 each proxy that sees the INVITE will also see all responses to the INVITE. When Alice's softphone receives the 180 (Ringing) response, it passes this information to Alice, perhaps using an audio ringback tone or by displaying a message on Alice's screen. In this example, Bob decides to answer the call. When he picks up the handset, his SIP phone sends a 200 (OK) response to indicate that the call has been answered. The 200 (OK) contains a message body with the SDP media description of the type of session that Bob is willing to establish with Alice. As a result, there is a two-phase exchange of SDP messages; Alice sent one to Bob, and Bob sent one back to Alice. This two-phase exchange provides basic negotiation capabilities and is based on a simple offer/answer model of SDP exchange. If Bob did not wish to answer the call or was busy on another call, an error response would have been sent instead of the 200 (OK), which would have resulted in no media session being established. The complete list of SIP response codes is in Section 25. The 200 (OK) (message F9 in Figure 1) might look like this as Bob sends it out: SIP/2.0 200 OK Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob ;tag=a6c85cf From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: Content-Type: application/sdp Content-Length: 131 (Bob's SDP not shown) The first line of the response contains the response code (200) and the reason phrase (OK). The remaining lines contain header fields. The Via header fields, To, From, Call- ID, and CSeq are all copied from the INVITE request. (There are three Via headers - one added by Alice's SIP phone, one added by the atlanta.com proxy, and one added by the biloxi.com proxy.) Bob's SIP phone has added a tag parameter to the To header field. This tag will be incorporated by both User Agents into the dialog and will be included in all future requests and responses in this call. The Contact header field contains a URI at which Bob can be directly reached at his SIP phone. The Content- Various Authors [Page 8] Internet Draft SIP January 28, 2002 Type and Content-Length refer to the message body (not shown) that contains Bob's SDP media information. In additon to DNS and location service lookups shown in this example, proxy servers can make flexible "routing decisions" to decide where to send a request. For example, if Bob's SIP phone returned a 486 (Busy Here) response, the biloxi.com proxy server could proxy the INVITE to Bob's voicemail server. A proxy server can also send an INVITE to a number of locations at the same time. This type of parallel search is known as "forking". In this case, the 200 (OK) is routed back through the two proxies and is received by Alice's softphone which then stops the ringback tone and indicates that the call has been answered. Finally, an acknowledgement message, ACK, is sent by Alice to Bob to confirm the reception of the final response (200 (OK)). In this example, the ACK is sent directly from Alice to Bob, bypassing the two proxies. This is because, through the INVITE/200 (OK) exchange, the two SIP user agents have learned each other's IP address through the Contact header fields, which was not known when the initial INVITE was sent. The lookups performed by the two proxies are no longer needed, so they drop out of the call flow. This completes the INVITE/200/ACK three-way handshake used to establish SIP sessions and is the end of the transaction. Full details on session setup are in Section 13. Alice and Bob's media session has now begun, and they send media packets using the format agreed to in the exchange of SDP. In general, the end-to-end media packets take a different path from the SIP signaling messages. During the session, either Alice or Bob may decide to change the characteristics of the media session. This is accomplished by sending a re-INVITE containing a new media description. If the change is accepted by the other party, a 200 (OK) is sent, which is itself responded to with an ACK. This re-INVITE references the existing dialog so the other party knows that it is to modify an existing session instead of establishing a new session. If the change is not accepted, an error response, such as a 406 (Not Acceptable), is sent, which also receives an ACK. However, the failure of the re-INVITE does not cause the existing call to fail - the session continues using the previously negotiated characteristics. Full details on session modification are in Section 14. At the end of the call, Bob disconnects (hangs up) first, and generates a BYE message. This BYE is routed directly to Alice's softphone, again bypassing the proxies. Alice confirms receipt of the BYE with a 200 (OK) response, which terminates the session and the BYE transaction. No ACK is sent - an ACK is only sent in response to Various Authors [Page 9] Internet Draft SIP January 28, 2002 a response to an INVITE request. The reasons for this special handling for INVITE will be discussed later, but relate to the reliability mechanisms in SIP, the length of time it can take for a ringing phone to be answered, and forking. For this reason, request handling in SIP is often classified as either INVITE or non- INVITE, referring to all other methods besides INVITE. Full details on session termination are in Section 15. Full details of all the messages shown in the example of Figure 1 are shown in Section 26.2. In some cases, it may be useful for proxies in the SIP signaling path to see all the messaging between the endpoints for the duration of the session. For example, if the biloxi.com proxy server wished to remain in the SIP messaging path beyond the initial INVITE, it would add to the INVITE a required routing header field known as Record- Route that contained a URI resolving to the proxy. This information would be received by both Bob's SIP phone and (due to the Record- Route header field being passed back in the 200 (OK)) Alice's softphone and stored for the duration of the dialog. The biloxi.com proxy server would then receive and proxy the ACK, BYE, and 200 (OK) to the BYE. Each proxy can independently decide to receive subsequent messaging, and that messaging will go through all proxies that elect to receive it. This capability is frequently used for proxies that are providing mid-call features. Registration is another common operation in SIP. Registration is one way that the biloxi.com server can learn the current location of Bob. Upon initialization, and at periodic intervals, Bob's SIP phone sends REGISTER messages to a server in the biloxi.com domain known as a SIP registrar. The REGISTER messages associate Bob's SIP URI (sip:bob@biloxi.com) with the machine he is currently logged in at (conveyed as a SIP URI in the Contact header). The registrar writes this association, also called a binding, to a database, called the location service , where it can be used by the proxy in the biloxi.com domain. Often, a registrar server for a domain is co- located with the proxy for that domain. It is an important concept that the distinction between types of SIP servers is logical, not physical. Bob is not limited to registering from a single device. For example, both his SIP phone at home and the one in the office could send registrations. This information is stored together in the location service and allows a proxy to perform various types of searches to locate Bob. Similarly, more than one user can be registered on a single device at the same time. The location service is just an abstract concept. It generally Various Authors [Page 10] Internet Draft SIP January 28, 2002 contains information that allows a proxy to input a URI and get back a translated URI that tells the proxy where to send the request. Registrations are one way to create this information, but not the only way. Arbitrary mapping functions can be programmed, at the discretion of the administrator. Finally, it is important to note that in SIP, registration is used for routing incoming SIP requests and has no role in authorizing outgoing requests. Authorization and authentication are handled in SIP either on a request-by-request, challenge/response mechanism, or using a lower layer scheme as discussed in Section 22. The complete set of SIP message details for this registration example is in Section 26.1. Additional operations in SIP, such as querying for the capabilities of a SIP server or client using OPTIONS, canceling a pending request using CANCEL, or supporting reliability of provisional responses using PRACK will be introduced in later sections. 5 Structure of the Protocol SIP is structured as a layered protocol, which means that its behavior is described in terms of a set of fairly independent processing stages with only a loose coupling between each stage. The protocol is structured into layers for the purpose of presentation and conciseness; it allows the grouping of functions common across elements into a single place. It does not dictate an implementation in any way. When we say that an element "contains" a layer, we mean it is compliant to the set of rules defined by that layer. Not every element specified by the protocol contains every layer. Furthermore, the elements specified by SIP are logical elements, not physical ones. A physical realization can choose to act as different logical elements, perhaps even on a transaction-by-transaction basis. The lowest layer of SIP is its syntax and encoding. Its encoding is specified using a BNF. The complete BNF is specified in Section 27. However, a basic overview of the structure of a SIP message can be found in Section 7. This section provides enough understanding of the format of a SIP message to facilitate understanding the remainder of the protocol. The next higher layer is the transport layer. This layer defines how a client takes a request and physically sends it over the network, and how a response is sent by a server and then received by a client. All SIP elements contain a transport layer. The transport layer is described in Section 19. Various Authors [Page 11] Internet Draft SIP January 28, 2002 The next higher layer is the transaction layer. Transactions are a fundamental component of SIP. A transaction is a request, sent by a client transaction (using the transport layer), to a server transaction, along with all responses to that request sent from the server transaction back to the client. The transaction layer handles application layer retransmissions, matching of responses to requests, and application layer timeouts. Any task that a UAC accomplishes takes place using a series of transactions. Discussion of transactions can be found in Section 17. User agents contain a transaction layer, as do stateful proxies. Stateless proxies do not contain a transaction layer. The transaction layer has a client component (referred to as a client transaction), and a server component (referred to as a server transaction), each of which are represented by an FSM that is constructed to process a particular request. The layer on top of the transaction layer is called the transaction user (TU), of which there are several types. When a TU wishes to send a request, it creates a client transaction instance and passes it the request along with the destination IP address, port, and transport to which to send the request. A TU which creates a client transaction can also cancel it. When a client cancels a transaction, it requests that the server stop further processing, revert to the state that existed before the transaction was initiated, and generate a specific error response to that transaction. This is done with a CANCEL request, which constitutes its own transaction, but references the transaction to be cancelled. Cancellation is described in Section 9. There are several different types of transaction users. A UAC contains a UAC core, a UAS contains a UAS core, and a proxy contains a proxy core. The behavior of the UAC and UAS cores depend largely on the method. However, there are some common rules for all methods. These rules are captured in Section 8. They primarily deal with construction of a request, in the case of a UAC, and processing of that request and generation of a response, in the case of a UAS. UAC and UAS core behavior for the REGISTER method is described in Section 10. Registrations play an important role in SIP. In fact, a UAS that handles a REGISTER is given a special name - a registrar - and it is described in that section. UAC and UAS core behavior for the OPTIONS method, used for determining the capabilities of a UA, are described in Section 11. Certain other requests are sent within a dialog. A dialog is a peer-to-peer SIP relationship between two user agents that persists Various Authors [Page 12] Internet Draft SIP January 28, 2002 for some time. The dialog facilitates sequencing of messages and proper routing of requests between the user agents. The INVITE method is the only way defined in this specification to establish a dialog. When a UAC sends a request that is within the context of a dialog, it follows the common UAC rules as discussed in Section 8, but also the rules for mid-dialog requests. Section 12 discusses dialogs and presents the procedures for their construction, and maintenance, in addition to construction of requests within a dialog. The UAS core can generate provisional responses to requests, which are responses that provide additional information about the request processing but do not indicate completion. Normally, provisional responses are not transmitted reliably. However, an optional mechanism exists for them to be transmitted reliably. This mechanism makes use of a method called PRACK, sent as a separate transaction within the dialog between the UAC and UAS, which is used to acknowledge a reliable provisional response. The most important method in SIP is the INVITE method, which is used to establish a session between participants. A session is a collection of participants, and streams of media between them, for the purposes of communication. Section 13 discusses how sessions are initiated, resulting in one or more SIP dialogs. Section 14 discusses how characteristics of that session are modified through the use of an INVITE request within a dialog. Finally, section 15 discusses how a session is terminated. The procedures of Sections 8, 10, 11, 12, 13, 14, and 15 deal entirely with the UA core (Section 9 describes cancellation, which applies to both UA core and proxy core). Section 16 discusses the proxy element, which facilitates routing of messages between user agents. 6 Definitions This specification uses a number of terms to refer to the roles played by participants in SIP communications. The terms and generic syntax of URI and URL are defined in RFC 2396 [9]. The following terms have special significance for SIP. Back-to-Back user agent: A back-to-back user agent (B2BUA) is a logical entity that receives a request and processes it as an user agent server (UAS). In order to determine how the request should be answered, it acts as an user agent client (UAC) and generates requests. Unlike a proxy server, it maintains dialog state and must participate in all requests sent on the dialogs it has established. Since it is a concatenation of a UAC and UAS, no explicit definitions are Various Authors [Page 13] Internet Draft SIP January 28, 2002 needed for its behavior. Call: A call is an informal term that refers to a dialog between peers generally set up for the purposes of a multimedia conversation. Call leg: Another name for a dialog. Call stateful: A proxy is call stateful if it retains state for a dialog from the initiating INVITE to the terminating BYE request. A call stateful proxy is always stateful, but the converse is not true. Client: A client is any network element that sends SIP requests and receives SIP responses. Clients may or may not interact directly with a human user. User agent clients and proxies are clients. Conference: A multimedia session (see below) that contains multiple participants. Dialog: A dialog is a peer-to-peer SIP relationship between a UAC and UAS that persists for some time. A dialog is established by SIP messages, such as a 2xx response to an INVITE request. A dialog is identified by a call identifier, local address, and remote address. A dialog was formerly known as a call leg in RFC 2543. Downstream: A direction of message forwarding within a transaction that refers to the direction that requests flow from the user agent client to user agent server. Final response: A response that terminates a SIP transaction, as opposed to a provisional response that does not. All 2xx, 3xx, 4xx, 5xx and 6xx responses are final. Header: A header is a component of a sip message that conveys information about the message. It is structured as a header name, followed by a colon, followed by its value. Home Domain: The domain providing service to a SIP user. Typically, this is the domain present in the URI in the address-of-record of a registration. Informational Response: Same as a provisional response. Initiator, calling party, caller: The party initiating a session (and dialog) with an INVITE request. A caller retains this Various Authors [Page 14] Internet Draft SIP January 28, 2002 role from the time it sends the initial INVITE which established a dialog, until the termination of that dialog. Invitation: An INVITE request. Invitee, invited user, called party, callee: The party that receives an INVITE request for the purposes of establishing a new session. A callee retains this role from the time it receives the INVITE until the termination of the dialog established by that INVITE. Location service: A location service is used by a SIP redirect or proxy server to obtain information about a callee's possible location(s). It contains a list of bindings of adress-of-record keys to zero or more contact addresses. The bindings can be created and removed in many ways; this specification defines a REGISTER method that updates the bindings. Loop: A request that arrives at a proxy, is forwarded, and later arrives back at the same proxy. When it arrives the second time, its Request-URI is identical to the first time, and other headers that affect proxy operation are unchanged, so that the proxy would make the same processing decision on the request it made the first time around. Looped requests are errors, and the procedures for detecting them and handling them are described by the protocol. Message: Data sent between SIP elements as part of the the protocol. SIP messages are either requests or responses. Method: The method is the primary function that a request is meant to invoke on a server. The method is carried in the request message itself. Example methods are INVITE and BYE. Outbound proxy: A proxy that receives all requests from a client, even though it is not the server resolved by the Request-URI. The outbound proxy sends these requests, after any local processing, to the address indicated in the Request-URI, or to another outbound proxy. Typically, a UA is manually configured with its outbound proxy, or can learn it through auto-configuration protocols. Parallel search: In a parallel search, a proxy issues several requests to possible user locations upon receiving an incoming request. Rather than issuing one request and then waiting for the final response before issuing the next request as in a sequential search , a parallel search Various Authors [Page 15] Internet Draft SIP January 28, 2002 issues requests without waiting for the result of previous requests. Provisional response: A response used by the server to indicate progress, but that does not terminate a SIP transaction. 1xx responses are provisional, other responses are considered final. Normally, provisional responses are not sent reliably. A provisional response that is sent reliably is referred to as a reliable provisional response Proxy, proxy server: An intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is passed on to another entity "closer" to the targeted user. Proxies are also useful for enforcing policy (for example, making sure a user is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it. Recursion: A client recurses on a 3xx response when it generates a new request to the URIs in the Contact headers in the response. Redirect Server: A redirect server is a server that generates 3xx responses to requests it receives, directing the client to contact an alternate URI. Registrar: A registrar is a server that accepts REGISTER requests, and places the information it receives in those requests into the location service for the domain it handles. Regular Transaction: A regular transaction is any transaction with a method other than INVITE, ACK, or CANCEL. Reliable Provisional Response: A provisional response that is sent reliably from the UAS to UAC. Request: A SIP message sent from a client to a server, for the purpose of invoking a particular operation. Response: A SIP message sent from a server to a client, for indicating the status of a request sent from the client to the server. Ringback: Ringback is the signaling tone produced by the calling party's application indicating that a called party is being Various Authors [Page 16] Internet Draft SIP January 28, 2002 alerted (ringing). Route Refresh Request: A route refresh request sent within a dialog is defined as a request that can modify the route set of the dialog. Server: A server is a network element that receives requests in order to service them and sends back responses to those requests. Examples of servers are proxies, user agent servers, redirect servers, and registrars. Sequential search: In a sequential search, a proxy server attempts each contact address in sequence, proceeding to the next one only after the previous has generated a non- 2xx final response. Session: From the SDP specification: "A multimedia session is a set of multimedia senders and receivers and the data streams flowing from senders to receivers. A multimedia conference is an example of a multimedia session." (RFC 2327 [5]) (A session as defined for SDP can comprise one or more RTP sessions.) As defined, a callee can be invited several times, by different calls, to the same session. If SDP is used, a session is defined by the concatenation of the user name , session id , network type , address type , and address elements in the origin field. (SIP) transaction: A SIP transaction occurs between a client and a server and comprises all messages from the first request sent from the client to the server up to a final (non-1xx) response sent from the server to the client, and the ACK for the response in the case the response was a non-2xx. The ACK for a 2xx response is a separate transaction. Spiral: A spiral is a SIP request that is routed to a proxy, forwarded onwards, and arrives once again at that proxy, but this time, differs in a way that will result in a different processing decision than the original request. Typically, this means that the request's Request-URI differs from its previous arrival. A spiral is not an error condition, unlike a loop. A typical cause for this is call forwarding. A user calls joe@example.com. The example.com proxy forwards it to Joe's PC, which in turn, forwards it to bob@example.com. This request is proxied back to the example.com proxy. However, this is not a loop. Since the request is targeted at a different user, it is considered a spiral, and is a valid condition. Various Authors [Page 17] Internet Draft SIP January 28, 2002 Stateful proxy: A logical entity that maintains the client and server transaction state machines defined by this specification during the processing of a request. Also known as a transaction stateful proxy. The behavior of a stateful proxy is further defined in Section 16. A stateful proxy is not the same as a call stateful proxy. Stateless proxy: A logical entity that does not maintain the client or server transaction state machines defined in this specification when it processes requests. A stateless proxy forwards every request it receives downstream and every response it receives upstream. Transaction User (TU): The layer of protocol processing that resides above the transaction layer. Transaction users include the UAC core, UAS core, and proxy core. Upstream: A direction of message forwarding within a transaction that refers to the direction that responses flow from the user agent server to user agent client. URL-encoded: A character string encoded according to RFC 1738, Section 2.2 [10]. User agent client (UAC): A user agent client is a logical entity that creates a new request, and then uses the client transaction state machinery to send it. The role of UAC lasts only for the duration of that transaction. In other words, if a piece of software initiates a request, it acts as a UAC for the duration of that transaction. If it receives a request later on, it assumes the role of a user agent server for the processing of that transaction. UAC Core: The set of processing functions required of a UAC that reside above the transaction and transport layers. User agent server (UAS): A user agent server is a logical entity that generates a response to a SIP request. The response accepts, rejects or redirects the request. This role lasts only for the duration of that transaction. In other words, if a piece of software responds to a request, it acts as a UAS for the duration of that transaction. If it generates a request later on, it assumes the role of a user agent client for the processing of that transaction. UAS Core: The set of processing functions required at a UAS that reside above the transaction and transport layers. Various Authors [Page 18] Internet Draft SIP January 28, 2002 User agent (UA): A logical entity that can act as both a user agent client and user agent server for the duration of a dialog. The role of UAC and UAS as well as proxy and redirect servers are defined on a transaction-by-transaction basis. For example, the user agent initiating a call acts as a UAC when sending the initial INVITE request and as a UAS when receiving a BYE request from the callee. Similarly, the same software can act as a proxy server for one request and as a redirect server for the next request. Proxy, location, and registrar servers defined above are logical entities; implementations MAY combine them into a single application. 7 SIP Messages SIP is a text-based protocol and uses the ISO 10646 character set in UTF-8 encoding (RFC 2279 [11]). A SIP message is either a request from a client to a server, or a response from a server to a client. Both Request (section 7.1) and Response (section 7.2) messages use the basic format of RFC 2822 [7], even though the syntax differs in character set and syntax specifics. (SIP allows header fields that would not be valid RFC 2822 header fields, for example.) Both types of messages consist of a start-line, one or more header fields (also known as "headers"), an empty line indicating the end of the header fields, and an optional message-body. generic-message = start-line *message-header CRLF [ message-body ] The start-line, each message-header line, and the empty line MUST be terminated by a carriage-return line-feed sequence (CRLF). Note that the empty line MUST be present even if the message-body is not. Except for the above difference in character sets, much of SIP's message and header field syntax is identical to HTTP/1.1. Rather than repeating the syntax and semantics here, we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1 specification (RFC 2616 [12]). However, SIP is not an extension of HTTP. Various Authors [Page 19] Internet Draft SIP January 28, 2002 7.1 Requests SIP requests are distinguished by having a Request-Line for a start- line. A Request-Line contains a method name, a Request-URI, and the protocol version separated by a single space (SP) character. The Request-Line ends with CRLF. No CR or LF are allowed except in the end-of-line CRLF sequence. No LWS is allowed in any of the elements. Method Request-URI SIP-Version Method: This specification defines seven methods: REGISTER for registering contact information, INVITE, ACK, PRACK and CANCEL for setting up sessions, BYE for terminating sessions and OPTIONS for querying servers about their capabilities. SIP extensions, documented in standards track RFCs, may define additional methods. Request-URI: The Request-URI is a SIP URI as described in Section 23.1 or a general URI (RFC 2396 [9]). It indicates the user or service to which this request is being addressed. The Request-URI MUST NOT contain unescaped spaces or control characters and MUST NOT be enclosed in "<>". SIP elements MAY support Request-URIs with schemes other than "sip", for example the "tel" URI scheme of RFC 2806 [13]. SIP elements MAY translate non-SIP URIs using any mechanism at their disposal, resulting in either a SIP URI or some other scheme. SIP-Version: Both request and response messages include the version of SIP in use, and follow [H3.1] (with HTTP replaced by SIP, and HTTP/1.1 replaced by SIP/2.0) regarding version ordering, compliance requirements, and upgrading of version numbers. To be compliant with this specification, applications sending SIP messages MUST include a SIP-Version of "SIP/2.0". The SIP-Version string is case-insensitive, but implementations MUST send upper- case. Unlike HTTP/1.1, SIP treats the version number as a literal string. In practice, this should make no difference. 7.2 Responses Various Authors [Page 20] Internet Draft SIP January 28, 2002 SIP responses are distinguished from requests by having a Status-Line as their start-line. A Status-Line consists of the protocol version followed by a numeric Status-Code and its associated textual phrase, with each element separated by a single SP character. No CR or LF is allowed except in the final CRLF sequence. SIP-version Status-Code Reason-Phrase The Status-Code is a 3-digit integer result code that indicates the outcome of an attempt to understand and satisfy a request. The Reason-Phrase is intended to give a short textual description of the Status-Code. The Status-Code is intended for use by automata, whereas the Reason-Phrase is intended for the human user. A client is not required to examine or display the Reason-Phrase. While this specification suggests specific wording for the reason phrase, implementations MAY choose other text, e.g., in the language indicated in the Accept-Language header field of the request. The first digit of the Status-Code defines the class of response. The last two digits do not have any categorization role. For this reason, any response with a status code between 100 and 199 is referred to as a "1xx response", any response with a status code between 200 and 299 as a "2xx response", and so on. SIP/2.0 allows six values for the first digit: 1xx: Provisional -- request received, continuing to process the request; 2xx: Success -- the action was successfully received, understood, and accepted; 3xx: Redirection -- further action needs to be taken in order to complete the request; 4xx: Client Error -- the request contains bad syntax or cannot be fulfilled at this server; 5xx: Server Error -- the server failed to fulfill an apparently valid request; 6xx: Global Failure -- the request cannot be fulfilled at any server. Section 25 defines these classes and describes the individual codes. 7.3 Header Fields SIP header fields are similar to HTTP header fields in both syntax Various Authors [Page 21] Internet Draft SIP January 28, 2002 and semantics. In particular, SIP header fields follow the [H4.2] definitions of syntax for message-header, the rules for extending header fields over multiple lines, the use of multiple message-header fields with the same field-name, and the rules regarding ordering of header fields. 7.3.1 Header Field Format Header fields follow the same generic header format as that given in Section 2.2 of RFC 2822 [7]. Each header field consists of a field name followed by a colon (":") and the field value. field-name: field-value The formal grammar for a message-header specified in Section 27 allows for an arbitrary amount of whitespace on either side of the colon; however, implementations should avoid spaces between the field name and the colon and use a single space (SP) between the colon and the field-value. Thus, Subject: lunch Subject : lunch Subject :lunch Subject: lunch are all valid and equivalent, but the last is the preferred form. Header fields can be extended over multiple lines by preceding each extra line with at least one SP or horizontal tab (HT). The line break and the whitespace at the beginning of the next line are treated as a single SP character. Thus, the following are equivalent: Subject: I know you're there, pick up the phone and talk to me! Subject: I know you're there, pick up the phone and talk to me! The relative order of header fields with different field names is not significant. However, it is RECOMMENDED that headers which are needed for proxy processing (Via, Route, Record-Route, Proxy-Require, Max-Forwards, and Proxy-Authorization, for example) appear towards the top of the message, to facilitate rapid parsing. The relative order of header fields with the same field name is important. Multiple header fields with the same field-name MAY be present in a message if and only if the entire field-value for that header field is defined as a comma-separated list (that is, #(values)). It MUST be Various Authors [Page 22] Internet Draft SIP January 28, 2002 possible to combine the multiple header fields into one "field-name: field-value" pair, without changing the semantics of the message, by appending each subsequent field-value to the first, each separated by a comma. Implementations MUST be able to process multiple header fields with the same name in any combination of the single-value-per-line or comma-separated value forms. The following groups of header fields are valid and equivalent: Route: Subject: Lunch Route: Route: Route: , Route: Subject: Lunch Subject: Lunch Route: , , Each of the following blocks is valid but not equivalent to the others: Route: Route: Route: Route: Route: Route: Route: ,, The format of a header field-value is defined per header-name. It will always be either an opaque sequence of TEXT-UTF8 octets, or a combination of whitespace, tokens, separators, and quoted strings. Many existing headers will adhere to the general form of a value followed by a semi-colon separated sequence of parameter-name, parameter-value pairs: field-name: field-value *(;parameter-name=parameter-value) Various Authors [Page 23] Internet Draft SIP January 28, 2002 Even though an arbitrary number of parameter pairs may be attached to a header field value, any given parameter-name MUST NOT appear more than once. All new header fields MUST follow this generic format unless they have been inherited from other RFC 2822-like specifications. When comparing header fields, field names are always case- insensitive. Unless otherwise stated in the definition of a particular header field, field values, parameter names, and parameter values are case-insensitive. Tokens are always case-insensitive. Unless specified otherwise, values expressed as quoted strings are case-sensitive. For example, Contact: ;expires=3600 is equivalent to CONTACT: ;ExPiReS=3600 and Content-Disposition: session;handling=optional is equivalent to content-disposition: Session;HANDLING=OPTIONAL The following two header fields are not equivalent: Warning: 370 devnull "Choose a bigger pipe" Warning: 370 devnull "CHOOSE A BIGGER PIPE" 7.3.2 Header Field Classification Some header fields only make sense in requests or responses. These are called request header fields and response header fields, respectively. If a header appears in a message not matching its category (such as a request header field in a response), it MUST be Various Authors [Page 24] Internet Draft SIP January 28, 2002 ignored. Section 24 defines the classification of each header field. 7.3.3 Compact Form SIP provides a mechanism to represent common header fields in an abbreviated form. This may be useful when messages would otherwise become too large to be carried on the transport available to it (exceeding the maximum transmission unit (MTU) when using UDP, for example). These compact forms are defined in Section 24. A compact form MAY be substituted for the longer form of a header name at any time without changing the semantics of the message. The same type of header field MAY appear in both long and short forms within the same message. Implementations MUST accept both the long and short forms of each header name. 7.4 Bodies Requests, including new requests defined in extensions to this specification, MAY contain message bodies unless otherwise noted. The interpretation of the body depends on the request method. For response messages, the request method and the response status code determine the type and interpretation of any message body. All responses MAY include a body. 7.4.1 Message Body Type The Internet media type of the message body MUST be given by the Content-Type header field. If the body has undergone any encoding such as compression, then this MUST be indicated by the Content- Encoding header field; otherwise, Content-Encoding MUST be omitted. If applicable, the character set of the message body is indicated as part of the Content-Type header-field value. The "multipart" MIME type defined in RFC 2046 [14] MAY be used within the body of the message. Implementations that send requests containing multipart message bodies MUST send a session description as a non-multipart message body if the remote implementation requests this through an Accept header field that does not contain multipart. Note that SIP messages MAY contain binary bodies or body parts. 7.4.2 Message Body Length The body length in bytes is provided by the Content-Length header field. Section 24.14 describes the necessary contents of this header in detail. Various Authors [Page 25] Internet Draft SIP January 28, 2002 The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP. (Note: The chunked encoding modifies the body of a message in order to transfer it as a series of chunks, each with its own size indicator.) 7.5 Framing SIP messages Unlike HTTP, SIP implementations can use UDP or other unreliable datagram protocols. Each such datagram carries one request or response. See Section 19 on constraints on usage of unreliable transports. Likewise, implementations processing SIP messages over stream- oriented transports MUST ignore any CRLF appearing before the start- line [H4.1] 8 General User Agent Behavior A user agent represents an end system. It contains a User Agent Client (UAC), which generates requests, and a User Agent Server (UAS) which responds to them. A UAC is capable of generating a request based on some external stimulus (the user clicking a button, or a signal on a PSTN line), and processing a response. A UAS is capable of receiving a request, and generating a response, based on user input, external stimulus, the result of a program execution, or some other mechanism. When a UAC sends a request, it will pass through some number of proxy servers, which forward the request towards the UAS. When the UAS generates a response, the response is forwarded towards the UAC. UAC and UAS procedures depend strongly on two factors. First, whether the request or response is inside or outside of a dialog, and second, based on the method of a request. Dialogs are discussed thoroughly in Section 12; they represent a peer-to-peer relationship between user agents, and are established by specific SIP methods, such as INVITE. In this section, we discuss the method independent rules for UAC and UAS behavior when processing requests that are outside of a dialog. This includes, of course, the requests which themselves establish a dialog. Security procedures for requests and responses outside of a dialog are described in Section 22. Specifically, mechanisms exist for the UAS and UAC to mutually authenticate. A limited set of privacy features are also supported through encryption of bodies using S/MIME. Various Authors [Page 26] Internet Draft SIP January 28, 2002 8.1 UAC Behavior This section covers UAC behavior outside of a dialog. 8.1.1 Generating the Request A valid SIP request formulated by a UAC MUST at a minimum contain the following headers: To, From, CSeq, Call-ID, Max-Forwards, and Via; all of these headers are mandatory in all SIP messages. These six headers are the fundamental building blocks of a SIP message, as they jointly provide for most of the critical message routing services including the addressing of messages, the routing of responses, limiting message propagation, ordering of messages, and the unique identification of transactions. These headers are in addition to the mandatory request line, which contains the method, Request-URI and SIP version. Examples of requests sent outside of a dialog include an INVITE to establish a session (Section 13) and an OPTIONS to query for capabilities (Section 11). 8.1.1.1 Request-URI The initial Request-URI of the message SHOULD be set to the value of the URI in the To field. One notable exception is the REGISTER method; behavior for setting the Request-URI of register is given in Section 10. Another exception is the case of pre-existing Route headers; in that case, the procedures of Section 12.2.1.1 as they pertain to the Request-URI are followed, even though there is no dialog. Pre- existing Route headers are an ordered set of URIs that identify a chain of servers to which outgoing requests from a UAC will be sent. Commonly, they are configured on the user agent by a user or service provider manually, or through some non-SIP mechanism. They are most often used to identify a local outbound proxy server through which a UAC will send all requests, which in turn allows service providers to maintain a common point of policy enforcement for requests. 8.1.1.2 To The To general-header field first and foremost specifies the desired "logical" recipient of the request, or the address of record of the user or resource that is the target of this request. This may or may not be the ultimate recipient of the request. The To header MAY contain a SIP URI, but it may also make use of other URI schemes (the tel URL [13], for example) when appropriate. All SIP implementations MUST support the SIP URI. The To header field allows for a display Various Authors [Page 27] Internet Draft SIP January 28, 2002 name. A UAC may learn how to populate the To header field for a particular request in a number of ways. Usually the user will suggest the To header field through a human interface, perhaps inputting the URI manually or selecting it from some sort of address book. Frequently, the user will not enter a complete URI, but rather, a string of digits or letters (i.e., "bob"). It is at the discretion of the UA to choose how to interpret this input. Using it to form the user part of a SIP URL implies that the UA wishes the name to be resolved in the domain the right hand side (RHS) of the at-sign in the SIP URI (i.e., sip:bob@example.com). The RHS will frequently be the home domain of the user, which allows for the home domain to process the outgoing request. This is useful for features like "speed dial" which require interpretation of the user part in the home domain. The tel URL is used when the UA does not wish to specify the domain that should interpret the user input. Rather, each domain that the request passes through would be given that opportunity. As an example, a user in an airport might log in, and send requests through an outbound proxy in the airport. If they enter "411" (this is the phone number for local directory assistance in the United States), that needs to be interpreted and processed by the outbound proxy in the airport, not the user's home domain. In this case, tel:411 would be the right choice. A request outside of a dialog MUST NOT contain a tag; the tag in the To field of a request identifies the peer of the dialog. Since no dialog is established, no tag is present. For further information on the To header see Section 24.41. The following is an example of valid To header: To: Carol 8.1.1.3 From The From general-header field indicates the logical identity of the initiator of the request, possibly the user's address of record. Like the To field, it contains a URI and optionally a display name. It is used by SIP elements to determine processing rules to apply to a request (for example, automatic call rejection). As such, it is very important that the From URI not contain IP addresses or the FQDN of the host the UA is running on, since these are not logical names. The From header field allows for a display name. A UAC SHOULD use the Various Authors [Page 28] Internet Draft SIP January 28, 2002 display name "Anonymous", along with a syntactically correct, but otherwise meaningless URI (like sip:988776a@ahhs.aa), if the identity of the client is to remain hidden. Usually the value that populates the From header field in requests generated by a particular user agent is pre-provisioned by the user or by the administrators of the user's local domain. If a particular user agent is used by multiple users, it might have switchable profiles that include a URI corresponding to the identity of the profiled user. Recipients of requests can authenticate the originator of a request in order to ascertain that they are who their From header field claims they are (see Section 20 for more on authentication). The From field MUST contain a new "tag" parameter, chosen by the UAC. See Section 23.3 for details on choosing a tag. For further information on the From header see Section 24.20. Examples: From: "Bob" ;tag=a48s From: sip:+12125551212@server.phone2net.com;tag=887s From: Anonymous ;tag=hyh8 8.1.1.4 Call-ID The Call-ID general-header field acts as a unique identifier to group together a series of messages. It MUST be the same for all requests and responses sent by either UA in a dialog. It SHOULD be the same in each registration from a UA. In a new request created by a UAC outside of any dialog, the Call-ID header MUST be selected by the UAC as a globally unique identifier over space and time unless overridden by method specific behavior. All SIP user agents must have a means to guarantee that the Call-ID headers they produce will not be inadvertently generated by any other user agent. Note that when requests are retried after certain failure responses that solicit an amendment to a request (for example, a challenge for authentication), these retried requests are not considered new requests, and therefore do not need new Call-ID headers; see Section 8.1.4.6. Use of cryptographically random identifiers [15] in the generation of Call-IDs is RECOMMENDED. Implementations MAY use the form Various Authors [Page 29] Internet Draft SIP January 28, 2002 "localid@host". Call-IDs are case-sensitive and are simply compared byte-by-byte. Using cryptographically random identifiers provides some protection against session hijacking and reduces the likelihood of unintentional Call-ID collisions. No provisioning or human interface is required for the selection of the Call-ID header field value for a request. For further information on the Call-ID header see Section 24.8. Example: Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com 8.1.1.5 CSeq The Cseq header serves as a way to identify and order transactions. It consists of a sequence number and a method. The method MUST match that of the request. For requests outside of a dialog, the sequence number value is arbitrary, but MUST be expressible as a 32-bit unsigned integer and MUST be less than 2**31. As long as it follows the above guidelines, a client may use any mechanism it would like to select CSeq header field values. Section 12.2.1.1 discusses construction of the CSeq for requests within a dialog. Example: CSeq: 4711 INVITE 8.1.1.6 Max-Forwards The Max-Forwards header serves to limit the number of hops a request can transit on the way to its destination. It consists of an integer that is decremented by one at each hop. If the Max-Forwards value reaches 0 before the request reaches its destination, it will be rejected with a 483 Too Many Hops error response. A UAC MUST insert a Max-Forwards header field into each request it Various Authors [Page 30] Internet Draft SIP January 28, 2002 originates with a value of 70. 8.1.1.7 Via The Via header is used to indicate the transport used for the transaction, and to identify the location where the response is to be sent. When the UAC creates a request, it MUST insert a Via into that request. The protocol and version in the header MUST be SIP and 2.0, respectively. The Via header it inserts MUST contain a branch parameter. This parameter is used to uniquely identify the transaction created by that request. This parameter is used by both the client, and the server. The branch parameter value MUST be unique across time for all requests sent by the UA. The exception to this rule is CANCEL. As discussed below, a CANCEL request will have the same value of the branch parameter as the request it cancels. The uniqueness property of the branch ID parameter, to facilitate its use as a transaction ID, was not part of RFC 2543 The branch ID inserted by an element compliant with this specification MUST always begin with the characters "z9hG4bK". These 7 characters are used as a magic cookie (7 is deemed sufficient to ensure that an older RFC 2543 implementation would not pick such a value), so that servers receiving the request can determine that the branch ID was constructed in the fashion described by this specification (i.e., globally unique). Beyond this requirement, the precise format of the branch token is implementation-defined. The Via header maddr, ttl, and sent-by components will be set when the request is processed by the transport layer (Section 19). Via processing for proxies is described in Sections 3 and sec:proxy- response-processing-via. 8.1.1.8 Contact The Contact header provides a SIP URI that can be used to contact that specific instance of the user agent for subsequent requests. The Contact header MUST be present in any request that can result in the establishment of a dialog. For the methods defined in this specification, that includes only the INVITE request. For these requests, the scope of the Contact is the dialog. That is, the Various Authors [Page 31] Internet Draft SIP January 28, 2002 Contact header refers to the URI at which the UA would like to receive requests, for requests that are part of that dialog only. Only a single URI MUST be present. For further information on the Contact header, see Section 24.10. 8.1.1.9 Supported and Require If the UAC supports extensions to SIP that can be applied by the server to the response, the UAC SHOULD include a Supported header in the request listing the option tags (Section 23.2) for those extensions. This includes support for reliability for provisional responses, which is an extension even though it is defined within this specification. The option tag for reliability of provisional responses is 100rel The option-tags listed MUST only refer to extensions defined in standards-track RFCs. This is to prevent servers from insisting that clients implement non-standard, vendor-defined features in order to receive service. Extensions defined by experimental and informational RFCs are explicitly excluded from usage with the Supported header in a request, since they too are often used to document vendor-defined extensions. If the UAC wishes to insist that a UAS understand an extension that the UAC will apply to the request in order to process the request, it MUST insert a Require header into the request listing the option tag for that extension. If the UAC wishes to apply an extension to the request and insist that any proxies that are traversed understand that extension, it MUST insert a Proxy-Require header into the request listing the option tag for that extension. As with the Supported header, the option-tags in the Require header MUST only refer to extensions defined in standards-track RFCs. A Require header in a request with the option tag 100rel means that the UAC wishes for all provisional responses to this request to be transmitted reliably. This header MUST NOT be present in any requests excepting INVITE, although extensions to SIP may allow its usage with other request methods. 8.1.1.10 Additional Message Components After a new request has been created, and the headers described above have been properly constructed, any additional optional headers are added, as are any headers specific to the method. SIP requests MAY contain a MIME-encoded message-body. Regardless of Various Authors [Page 32] Internet Draft SIP January 28, 2002 the type of body that a request contains, certain headers must be formulated to characterize the contents of the body. For further information on these headers see Sections 24.14, 24.15 and 24.12. 8.1.2 Sending the Request The destination for the request is then computed. A loose-routing element MAY use local policy to determine the IP address, port, and transport used to reach the destination. One example of such a policy is an element configured to send requests to a default outbound proxy. Section 8.1.3 discusses restrictions on loose-routing policies. For other elements, the destination can be determined by applying the DNS proceedures described in [8] to the Request-URI. These procedures yield an ordered set of address, port, and transports to attempt. The UAC SHOULD follow the procedures defined there for stateful elements, trying each address until a server is contacted. Each try constitutes a new transaction, and therefore each carries a different Via header with a new branch parameter. Furthermore, the transport value in the Via header is set to whatever transport was determined for the target server. 8.1.3 Loose Routing Policies An element MAY apply a local loose-routing policy when preparing and sending a request. This policy MAY affect the Request-URI and Route header field values in the request as well as where the request is sent, and what transport mechanism is used to send it. Elements SHOULD use the strict-routing policy of removing the topmost value from a route set, placing it in the Request-URI and sending the request to the location indicated by that URI. This is the behavior of elements implementing earlier strict versions of Route/Record-Route. Where appropriate, elements MAY deviate from the strict-routing policy as long as the following restrictions are met: 8.1.3.1 Modifying the Route header field A loose-routing element MAY remove the topmost Route header field value. It MUST remove the topmost Route header field value if that value indicates a resource this element is responsible for. The element MUST NOT modify or remove any subsequent Route header field values. The element MAY place additional Route header field values into the Route header field before any existing values (effectivly pushing values onto the top of the Route set). Various Authors [Page 33] Internet Draft SIP January 28, 2002 A loose-routing element may chose to not remove the first Route header field value. For example, elements configured to use default outbound proxies in liu of using the DNS resolution proceedures will leave the topmost Route header field value in the message. When the topmost Route header field value indicates a resource this element is responsible for, the message has reached the element indicated by the route, and that value must be removed from the Route header field. This assures that Route header field values are consumed when the destination they indicate has been reached. 8.1.3.2 Modifying the Request-URI If the Request-URI identifies a resource for which this element is responsible, the loose-route policy SHOULD include modifying the Request-URI before sending the request. This restriction ensures that a Request-URI is modified once the resource it indicates has been reached. 8.1.3.3 Destination Choice A loose-routing policy MUST direct the request to or the resource indicated in the first Route header field value, or to a proxy it trusts to ensure this property. This restriction ensures the resource indicated by the topmost Route header field value is actually visited. 8.1.3.4 Loop Avoidance The Request-URI of a request emitted by a loose-routing element MUST differ from the URI in the first Route header field value. This restriction is necessary to avoid triggering false loop detections in older systems. The following algorithm can be applied to ensure sufficient difference in otherwise matching Request-URIs and first Route header field values. For each of these items, D is the address of the next hop (which may or may not be equivalent to A). If the topmost element in the received Route header field is Various Authors [Page 34] Internet Draft SIP January 28, 2002 , the outgoing request will contain METHOD sip:a@A;maddr=D Route: If the topmost element in the received Route header field is , the outgoing request will contain METHOD sip:a@A Route: If the topmost element in the received Route header field is and D!=B, the outgoing request will contain METHOD sip:a@A;maddr=D Route: 8.1.4 Processing Responses Responses are first processed by the transport layer and then passed up to the transaction layer. The transaction layer performs its processing and then passes it up to the TU. The majority of response processing in the TU is method specific. However, there are some general behaviors independent of the method. 8.1.4.1 Transaction Layer Errors In some cases, the response returned by the transaction layer will not be a SIP message, but rather a transaction layer event. The only event that the TU will encounter is the timeout event. When the timeout event is received from the transaction layer, it MUST be treated as if a 408 (Request Timeout) status code has been received. 8.1.4.2 Unrecognized Responses A UAC MUST treat any response it does not recognize as being equivalent to the x00 response code of that class, and MUST be able to process the x00 response code for all classes. For example, if a UAC receives an unrecognized response code of 431, it can safely assume that there was something wrong with its request and treat the response as if it had received a 400 (Bad Request) response code. Various Authors [Page 35] Internet Draft SIP January 28, 2002 8.1.4.3 Vias If more than one Via header field is present in a response, the UAC SHOULD discard the message. The presence of additional Via header fields that precede the originator of the request suggests that the message was misrouted or possibly corrupted. 8.1.4.4 Processing Reliable 1xx Responses A 1xx response that contains a Require header with the option tag 100rel is a reliable provisional response. The UA core follows the procedures in Section 18.2 to process the response, which will result in the generation of a PRACK request to acknowledge the reliable provisional response. 8.1.4.5 Processing 3xx responses Upon receipt of a redirection response (for example, a 3xx response status code), clients SHOULD use the URI(s) in the Contact header field to formulate one or more new requests based on the redirected request. If more than one URI is present in Contact header fields within the 3xx response, the UA MUST determine an order in which these contact addresses should be processed. UAs MUST consult the "q" parameter value of the Contact header fields (see Section 22.10) if available. Contact addresses MUST be ordered from highest qvalue to lowest. If no qvalue is present, a contact address is considered to have a qvalue of 1.0. Note that two or more contact addresses might have an equal qvalue - these URIs are eligible to be tried in parallel. Once an ordered list has been established, UACs MUST try to contact each URI in the ordered list in turn until a server responds. If there are contact addresses with an equal qvalue, the UAC MAY decide randomly on an order in which to process these addresses, or it MAY attempt to process contact addresses of equal qvalue in parallel. Note that for example, the UAC may effectively divide the ordered list into groups, processing the groups serially and processing the destinations in each group in parallel. If contacting an address in the list results in a failure, as defined in the next paragraph, the element moves to the next address in the list, until the list is exhausted. If the list is exhausted, then the request has failed. Various Authors [Page 36] Internet Draft SIP January 28, 2002 Failures SHOULD be detected through failure response codes (codes greater than 399) or network timeouts. Client transaction will report any transport layer failures to the transaction user. When a failure for a particular contact address is recieved, the client SHOULD try the next contact address. This will involve creating a new client transaction to deliver a new request. In order to create a request based on a contact address in a 3xx response, a UAC MUST copy the entire URI from the Contact header into the Request-URI, except for the "method-param" and "header" URI parameters (see Section 23.1.1 for a definition of these parameters). It uses the "header" parameters to create headers for the new request, overwriting headers associated with the redirected request in accordance with the guidelines in Section 23.1.5. Note that in some instances, headers that have been communicated in the contact address may instead append to existing request headers in the original redirected request. As a general rule, if the header can accept a comma-separated list of values, then the new header value MAY be appended to any existing values in the original redirected request. If the header does not accept multiple values, the value in the original redirected request MAY be overwritten by the header value communicated in the contact address. For example, if a contact address is returned with the following value: sip:user@host?Subject=foo&Call-Info= Then any Subject header in the original redirected request is overwritten, but the HTTP URL is merely appended to any existing Call-Info header field values. It is RECOMMENDED that the UAC reuse the same To, From, and Call-ID used in the original redirected request, but the UAC MAY also choose to update for example the Call-ID header field value for new requests. Finally, once the new request has been constructed, it is sent using a new client transaction, and therefore MUST have a new branch ID in the top Via field as discussed in Section 8.1.1.7. In all other respects, requests sent upon receipt of a redirect response SHOULD re-use the headers and bodies of the original Various Authors [Page 37] Internet Draft SIP January 28, 2002 request. In some instances, Contact header values may be cached at UAC temporarily or permanently depending on the status code received and the presence of an expiration interval; see Sections 25.3.2 and 25.3.3. 8.1.4.6 Processing 4xx responses Certain 4xx response codes require specific UA processing, independent of the method. If a 401 (Unauthorized) or 407 (Proxy Authentication Required) response is received, the UAC SHOULD follow the authorization procedures of Section 20.2 and Section 20.3 to retry the request with credentials. If a 413 (Request Entity Too Large) response is received (Section 25.4.11), the request contained a body that was longer than the UAS was willing to accept. If possible, the UAC SHOULD retry the request, either omitting the body or using one of a smaller length. If a 415 (Unsupported Media Type) response is received (Section 25.4.13), the request contained media types not supported by the UAS. The UAC SHOULD retry sending the request, this time only using content with types listed in the Accept header in the response, with encodings listed in the Accept-Encoding header in the response, and with languages listed in the Accept-Language in the response. If a 416 (Unsupported URI Scheme) response is received (Section 25.4.14, the Request-URI used a URI scheme not supported by the server. The client SHOULD retry the request, this time, using a SIP URI. If a 420 (Bad Extension) response is received (Section 25.4.15), the request contained a Require or Proxy-Require header listing an option-tag for a feature not supported by a proxy or UAS. The UAC SHOULD retry the request, this time omitting any extensions listed in the Unsupported header in the response. In all of the above cases, the request is retried by creating a new request with the appropriate modifications. This new request SHOULD have the same value of the Call-ID, To, and From of the previous request, but the CSeq should contain a new sequence number that is one higher than the previous. With other 4xx responses, including those yet to be defined, a retry may or may not be possible depending on the method and the use Various Authors [Page 38] Internet Draft SIP January 28, 2002 case. 8.2 UAS Behavior When a request outside of a dialog is processed by a UAS, there is a set of processing rules which are followed, independent of the method. Section 12 gives guidance on how a UAS can tell whether a request is inside or outside of a dialog. Note that request processing is atomic. If a request is accepted, all state changes associated with it MUST be performed. If it is rejected, all state changes MUST NOT be performed. 8.2.1 Method Inspection Once a request is authenticated (or no authentication was desired), the UAS MUST inspect the method of the request. If the UAS does not support the method of a request it MUST generate a 405 (Method Not Allowed) response. Procedures for generation of responses are described in Section 8.2.6. The UAS MUST also add an Allow header to the 405 (Method Not Allowed) response. The Allow header field MUST list the set of methods supported by the UAS generating the message. The Allow header field is presented in Section 24.5. If the method is one supported by the server, processing continues. 8.2.2 Header Inspection If a UAS does not understand a header field in a request (that is, the header is not defined in this specification or in any supported extension), the server MUST ignore that header and continue processing the message. A UAS SHOULD ignore any malformed headers that are not necessary for processing requests. 8.2.2.1 To and Request-URI The To header field identifies the original recipient of the request designated by the user identified in the From field. The original recipient may or may not be the UAS processing the request, due to call forwarding or other proxy operations. A UAS MAY apply any policy it wishes in determination of whether to accept requests when the To field is not the identity of the UAS. However, it is RECOMMENDED that a UAS accept requests even if they do not recognize the URI scheme (for example, a tel: URI) in the To header, or if the To header field does not address a known or current user of this UAS. If, on the other hand, the UAS decides to reject the request, it SHOULD generate a response with a 403 (Forbidden) status code and pass it to the Various Authors [Page 39] Internet Draft SIP January 28, 2002 server transaction layer for transmission. However, the Request-URI identifies the UAS that is to process the request. If the Request-URI uses a scheme not supported by the UAS, it SHOULD reject the request with a 416 (Unsupported URI Scheme) response. If the Request-URI does not identify an address that the UAS is willing to accept requests for, it SHOULD reject the request with a 404 (Not Found) response. Typically, a UA that uses the REGISTER method to bind its address of record to a specific contact address will see requests whose Request-URI equals those contact addressess. Other potential sources of received Request-URIs include the Contact headers of requests and responses sent by the UA that establish or refresh dialogs. 8.2.2.2 Merged Requests If the request has no tag in the To, the TU checks ongoing transactions. If the To, From, Call-ID, CSeq exactly match (including tags) those of any request received previously, but the branch-ID in the topmost Via is different from those received previously, the TU SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction. The same request has arrived at the UAS more than once, following different paths, most likely due to forking. The UAS processes the first such request received and responds with a 482 (Loop Detected) to the rest of them. 8.2.2.3 Require Assuming the UAS decides that it is the proper element to process the request, it examines the Require header field, if present. The Require general-header field is used by a UAC to tell a UAS about SIP extensions that the UAC expects the UAS to support in order to process the request properly. Its format is described in Section 24.33. If a UAS does not understand an option-tag listed in a Require header field, it MUST respond by generating a response with status code 420 (Bad Extension). The UAS MUST add an Unsupported header field, and list in it those options it does not understand amongst those in the Require header of the request. Upon receipt of the 420 (Bad Extension) the client SHOULD retry the request, this time without using those extensions listed in the Unsupported header field in the response. Note that Require and Proxy-Require MUST NOT be used in a SIP CANCEL request, or in an ACK request sent for a non-2xx response. These headers should be ignored if they are present in these requests. Various Authors [Page 40] Internet Draft SIP January 28, 2002 An ACK request for a 2xx response MUST contain only those Require and Proxy-Require values that were present in the initial request. Example: UAC->UAS: INVITE sip:watson@bell-telephone.com SIP/2.0 Require: 100rel UAS->UAC: SIP/2.0 420 Bad Extension Unsupported: 100rel This is to make sure that the client-server interaction will proceed without delay when all options are understood by both sides, and only slow down if options are not understood (as in the example above). For a well-matched client-server pair, the interaction proceeds quickly, saving a round-trip often required by negotiation mechanisms. In addition, it also removes ambiguity when the client requires features that the server does not understand. Some features, such as call handling fields, are only of interest to end systems. 8.2.3 Content Processing Assuming the UAS understands any extensions required by the client, the UAS examines the body of the message, and the headers that describe it. If there are any bodies whose type (indicated by the Content-Type), language (indicated by the Content-Language) or encoding (indicated by the Content-Encoding) are not understood, and that body part is not optional (as indicated by the Content- Disposition header), the UAS MUST reject the request with a 415 (Unsupported Media Type) response. The response MUST contain an Accept header listing the types of all bodies it understands, in the event the request contained bodies of types not supported by the UAS. If the request contained content encodings not understood by the UAS, the response MUST contain an Accept-Encoding header listing the encodings understood by the UAS. If the request contained content with languages not understood by the UAS, the response MUST contain an Accept-Language header indicating the languages understood by the UAS. Beyond these checks, body handling depends on the method and type. For further information on the processing of Content-specific headers Various Authors [Page 41] Internet Draft SIP January 28, 2002 see Section 7.4 as well as Section 24.11 through 24.15. 8.2.4 Applying Extensions A UAS that wishes to apply some extension when generating the response MUST only do so if support for that extension is indicated in the Supported header in the request. If the desired extension is not supported, the server SHOULD rely only on baseline SIP and any other extensions supported by the client. To ensure that the SHOULD can be fulfilled, any specification of a new extension MUST include discussion of how to return gracefully to baseline SIP when the extension is not present. In rare circumstances, where the server cannot process the request without the extension, the server MAY send a 421 (Extension Required) response. This response indicates that the proper response cannot be generated without support of a specific extension. The needed extension(s) MUST be included in a Require header in the response. This behavior is NOT RECOMMENDED, as it will generally break interoperability. Any extensions applied to a non-421 response MUST be listed in a Require header included in the response. Of course, the server MUST NOT apply extensions not listed in the Supported header in the request. As a result of this, the Require header in a response will only ever contain option tags defined in standards-track RFCs. 8.2.5 Processing the Request Assuming all of the checks in the previous subsections are passed, the UAS processing becomes method-specific. Section 10 covers the REGISTER request, section 11 covers the OPTIONS request, section 13 covers the INVITE request, and section 15 covers the BYE request. 8.2.6 Generating the Response When a UAS wishes to construct a response to a request, it follows these procedures. Additional procedures may be needed depending on the status code of the response and the circumstances of its construction. These additional procedures are documented elsewhere. 8.2.6.1 Sending a Provisional Response One largely non-method-specific guideline for the generation of responses is that UASs SHOULD NOT issue a provisional response for a non-INVITE request. Rather, UASs SHOULD generate a final response to a non-INVITE request as sooon as possible. When a 100 (Trying) response is generated, any Timestamp header present in the request MUST be copied into this 100 (Trying) Various Authors [Page 42] Internet Draft SIP January 28, 2002 response. 8.2.6.2 Headers and Tags The From field of the response MUST equal the From field of the request. The Call-ID field of the response MUST equal the Call-ID field of the request. The Cseq field of the response MUST equal the Cseq field of the request. The Via headers in the response MUST equal the Via headers in the request and MUST maintain the same ordering. If a request contained a To tag in the request, the To field in the response MUST equal that of the request. However, if the To field in the request did not contain a tag, the URI in the To field in the response MUST equal the URI in the To field in the request; additionally, the UAS MUST add a tag to the To field in the response (with the exception of the 100 (Trying) response, in which a tag MAY be present). This serves to identify the UAS that is responding, possibly resulting in a component of a dialog ID. The same tag MUST be used for all responses to that request, both final and provisional (again excepting the 100 (Trying)). Procedures for generation of tags are defined in Section 23.3. 8.2.7 Stateless UAS Behavior A stateless UAS is a UAS that does not maintain transaction state. It replies to requests normally, but discards any state that would ordinarily be retained by a UAS after a response has been sent. If a stateless UAS receives a retransmission of a request, it regenerates the response and resends it, just as if it were the replying to the first instance of the request. Stateless UASs do not use a transaction layer; they receive requests directly from the transport layer amd send responses directly to the transport layer. The stateless UAS role is needed primarily to handle unauthenticated requests for which a challenge response is issued. If unauthenticated requests were handled statefully, then malicious floods of unauthenticated requests could create massive amounts of transaction state that might slow or complete halt call processing in a UAS, effectively creating a denial of service condition; for more information see Section 22.1.5. The most important behaviors of a stateless UAS are the following: o A stateless UAS MUST NOT send provisional (1xx) responses. o A stateless UAS MUST NOT retransmit responses. o A stateless UAS MUST ignore ACK requests. Various Authors [Page 43] Internet Draft SIP January 28, 2002 o A stateless UAS MUST ignore CANCEL requests. o To header tags MUST be generated for responses in a stateless manner - in a manner that will generate the same tag for the same request consistently. For information on tag construction see Section 23.3. In all other respects, a stateless UAS behaves in the same manner as a stateful UAS. A UAS can operate in either a stateful or stateless mode for each new request. 8.3 Redirect Servers In some architectures it may be desirable to reduce the processing load on proxy servers that are responsible for routing requests, and improve signaling path robustness, by relying on redirection. Redirection allows servers to push routing information for a request back in a response to the client, thereby taking themselves out of the loop of further messaging for this transaction while still aiding in locating the target of the request. When the originator of the request receives the redirection, it will send a new request based on the URI it has received. By propagating URIs from the core of the network to its edges, redirection allows for considerable network scalability. A redirect server is logically constituted of a server transaction layer and a transaction user that has access to a location service of some kind (see Section 10 for more on registrars and location services). This location service is effectively a database containing mappings between a single URI and a set of one or more alternative locations at which the target of that URI can be found. A redirect server does not issue any SIP requests of its own. After receiving a request other than CANCEL, the server gathers the list of alternative locations from the location service and either returns a final response of class 3xx or it refuses the request. For well- formed CANCEL requests, it SHOULD return a 2xx response. This response ends the SIP transaction. The redirect server maintains transaction state for an entire SIP transaction. It is the responsibility of clients to detect forwarding loops between redirect servers. When a redirect server returns a 3xx response to a request, it populates the list of (one or more) alternative locations into Contact headers. An "expires" parameter to the Contact header may also be supplied to indicate the lifetime of the Contact data. The Contact header field contains URIs giving the new locations or Various Authors [Page 44] Internet Draft SIP January 28, 2002 user names to try, or may simply specify additional transport parameters. A 301 (Moved Permanently) or 302 (Moved Temporarily) response may also give the same location and username that was targeted by the initial request but specify additional transport parameters such as a different server or multicast address to try, or a change of SIP transport from UDP to TCP or vice versa. However, redirect servers MUST NOT redirect a request to a URI equal to the one in the Request-URI; instead, provided that the URI does not point to itself, the redirect server SHOULD proxy the request to the destination URI. If a client is using an outbound proxy, and that proxy actually redirects requests, a potential arises for infinite redirection loops. Note that the Contact header field MAY also refer to a different entity than the one originally called. For example, a SIP call connected to GSTN gateway may need to deliver a special informational announcement such as "The number you have dialed has been changed." A Contact response header field can contain any suitable URI indicating where the called party can be reached, not limited to SIP URIs. For example, it could contain URIs for phones, fax, or irc (if they were defined) or a mailto: (RFC 2368, [16]) URL. The "expires" parameter of the Contact header field indicates how long the URI is valid. The value of the parameter is a number indicating seconds. If this parameter is not provided, the value of the Expires header field determines how long the URI is valid. Implementations MAY treat values larger than 2**32-1 (4294967295 seconds or 136 years) as equivalent to 2**32-1. Malformed values should be treated as equivalent to 3600. Redirect servers MUST ignore features that are not understood (including unrecognized headers, Required extensions, or even method names) and proceed with the redirection of the session in question. If a particular extension requires that intermediate devices support it, the extension MUST be tagged in the Proxy-Require field as well (see Section 24.29). 9 Canceling a Request The previous section has discussed general UA behavior for generating requests, and processing responses, for requests of all methods. In this section, we discuss a general purpose method, called CANCEL. The CANCEL request, as the name implies, is used to cancel a previous Various Authors [Page 45] Internet Draft SIP January 28, 2002 request sent by a client. Specifically, it asks the UAS to cease processing the request and to generate an error response to that request. CANCEL has no effect on a request to which a UAS has already responded. Because of this, it is most useful to CANCEL requests to which can take a long time to respond. For this reason, CANCEL is most useful for INVITE requests, which can take a long time to generate a response. In that usage, a UAS that receives a CANCEL request for an INVITE, but has not yet sent a response, would "stop ringing", and then respond to the INVITE with a specific error response (a 487). CANCEL requests can be constructed and sent by any type of client, including both proxies and user agent clients. Section 15 discusses under what conditions a UAC would CANCEL an INVITE request, and Section 16.9 discusses proxy usage of CANCEL. Because a stateful proxy can generate its own CANCEL, a stateful proxy also responds to a CANCEL, rather than simply forwarding a response it would receive from a downstream element. For that reason, CANCEL is referred to as a "hop-by-hop" request, since it is responded to at each stateful proxy hop. 9.1 Client Behavior A CANCEL request SHOULD NOT be sent to cancel a request other than INVITE. Since requests other than INVITE are responded to immediately, sending a CANCEL for a non-INVITE request would always create a race condition. The following procedures are used to construct a CANCEL request. The Request-URI, Call-ID, To, the numeric part of CSeq and From header fields in the CANCEL request MUST be identical to those in the request being cancelled, including tags. A CANCEL constructed by a client MUST have only a single Via header, whose value matches the top Via in the request being cancelled. Using the same values for these headers allows the CANCEL to be matched with the request it cancels (Section 9.2 indicates how such matching occurs). However, the method part of the Cseq header MUST have a value of CANCEL. This allows it to be identified and processed as a transaction in its own right (See Section 17). If the request being cancelled contains Route header fields, the CANCEL request MUST include these Route header fields. This is needed so that stateless proxies are able to route CANCEL requests properly. Various Authors [Page 46] Internet Draft SIP January 28, 2002 The CANCEL request MUST NOT contain any Require or Proxy-Require header fields. Once the CANCEL is constructed, the client SHOULD check whether any response (provisional or final) has been received for the request being cancelled (herein referred to as the "original request"). The CANCEL request MUST NOT be sent if no provisional response has been received, rather, the client MUST wait for the arrival of a provisional response before sending the request. If the original request has generated a final response, the CANCEL SHOULD NOT be sent, as it is an effective no-op, since CANCEL has no effect on requests that have already generated a final response. When the client decides to send the CANCEL, it creates a client transaction for the CANCEL and passes it the CANCEL request along with the destination address, port, and transport. The destination address, port, and transport for the CANCEL MUST be identical to those used to send the original request. If it was allowed to send the CANCEL before receiving a response for the previous request, the server could receive the CANCEL before the original request. Note that both the transaction corresponding to the original request and the CANCEL transaction will complete independently. However, a UAC canceling a request cannot rely on receiving a 487 (Request Terminated) response for the original request, as an RFC 2543- compliant UAS will not generate such a response. If there is no final response for the original request in 64*T1 seconds (T1 is defined in Section 17.1.1.1), the client SHOULD then consider the original transaction cancelled and SHOULD destroy the client transaction handling the original request. 9.2 Server Behavior The CANCEL method requests that the TU at the server side cancel a pending transaction. The transaction to be canceled is determined by taking the CANCEL request, and then assuming that the request method were anything but CANCEL, apply the transaction matching procedures of Section 17.2.3. The matching transaction is the one to be canceled. The processing of a CANCEL request at a server depends on the type of server. A stateless proxy will forward it, a stateful proxy might respond to it and generate some CANCEL requests of its own, and a UAS will respond to it. See Section 16.9 for proxy treatment of CANCEL. A UAS first processes the CANCEL request according to the general UAS Various Authors [Page 47] Internet Draft SIP January 28, 2002 processing described in Section 8.2. However, since CANCEL requests are hop-by-hop and cannot be resubmitted, they cannot be challenged by the server in order to get proper credentials in an Authorization header field. Note also that CANCEL requests do not contain Require header fields. If the CANCEL did not find a matching transaction according to the procedure above, the CANCEL SHOULD be responded to with a 481 (Call Leg/Transaction Does Not Exist). If the transaction for the original request still exists, the behavior of the UAS on receiving a CANCEL request depends on whether it has already sent a final response for the original request. If it has, the CANCEL request has no effect on the processing of the original request, no effect on any session state, and no effect on the responses generated for the original request. If the UAS has not issued a final response for the original request, its behavior depends on the method of the original request. If the original request was an INVITE, the UAS SHOULD immediately respond to the INVITE with a 487 (Request Terminated). The behavior upon reception of a CANCEL request for any other method defined in this specification is effectively no-op. Extensions to this specification that define new methods MUST define the behavior of a UAS upon reception of a CANCEL for those methods. Regardless of the method of the original request, as long as the CANCEL matched an existing transaction, the CANCEL request itself is answered with a 200 (OK) response. This response is constructed following the procedures described in Section 8.2.6 noting that the To tag of the response to the CANCEL and the To tag in the response to the original request SHOULD be the same. The response to CANCEL is passed to the server transaction for transmission. 10 Registrations 10.1 Overview SIP offers a discovery capability. If a user wants to initiate a session with another user, SIP must discover the current host(s) that the destination user is reachable at. This discovery process is accomplished by SIP proxy servers, which are responsible for receiving a request, determining where to send it based on knowledge of the location of the user, and then sending it there. To do this, proxies consult an abstract service known as a location service , which provides address bindings for a particular domain. These address bindings map an incoming SIP URI, sip:bob@Biloxi.com , for example, to one or more SIP URIs which are somehow "closer" to the desired user, sip:bob@engineering.Biloxi.com , for example. Ultimately, a proxy will consult a location service which maps a received URI to the current host(s) that a user is logged in to. Various Authors [Page 48] Internet Draft SIP January 28, 2002 Registration creates bindings in a location service for a particular domain that associate an address-of-record URI with one or more contact addresses. This means that when a proxy for that domain receives a request whose request URI matches the address-of-record, the proxy will forward the request to the contact addresses registered to that address-of-record. Generally, it only makes sense to register an address-of-record at a location service for a domain when requests for that address-of-record would be routed to that domain. In most cases, this means that the domain of the registration will need to match the domain in the URI of the address-of-record. There are many ways by which the contents of the location service can be established. One way is administratively. In the above example, Bob is known to be a member of the engineering department through access to a corporate database. SIP provides a mechanism, however, for a user agent to explicitly create a binding. This mechanism is known as registration. Registration entails sending a REGISTER request to a special type of UAS known as a registrar. The registrar acts as a front end to the location service for a domain, reading and writing mappings based on the contents of the REGISTER requests. This location service will then be consulted by a proxy server that is responsible for routing requests for that domain. SIP does not mandate a particular mechanism for implementing the location service. The only requirement is that a registrar for some domain MUST be able to read and write data to the location service, and a proxy for that domain MUST be capable of reading that same data. A registrar MAY be co-located with a particular SIP proxy server for the same domain. 10.2 Constructing the REGISTER Request REGISTER requests add, remove and query bindings. A REGISTER request may add a new binding between an address-of-record and one or more contact addresses. Registration on behalf of a particular address- of-record may be performed by a suitably authorized third party. A client may also remove previous bindings, or query to determine which bindings are currently in place for an address-of-record. Except as noted, the construction of the REGISTER request and the behavior of clients sending a REGISTER request is identical to the general UAC behavior described in Section 8.1 and Section 17.1. The following header fields MUST be included: Request-URI: The Request-URI names the domain of the location Various Authors [Page 49] Internet Draft SIP January 28, 2002 service that the registration is meant for (e.g., "sip:chicago.com"). The "userinfo" and "@" components of the SIP URI MUST NOT be present. To: The To header field contains the address of record whose registration is to be created, queried or modified. The To header field and the Request-URI field typically differ, as the former contains a user name. This address-of-record MUST be a SIP URI. From: The From header field contains the address-of-record of the person responsible for the registration. The value is the same as the To header field unless the request is a third-party registration. Call-ID: All registrations from a user agent client SHOULD use the same Call-ID header value for registrations sent to a particular registrar. If the same client were to use different Call-ID values, a registrar could not detect whether a delayed REGISTER request might have arrived out of order. CSeq: The CSeq value guarantees proper ordering of REGISTER requests. A UA MUST increment the CSeq value by one for each REGISTER request with the same Call-ID. Contact: REGISTER requests contain zero or more Contact header fields, containing address bindings. User agents MUST NOT send a new registration (i.e., containing new Contact header fields, as opposed to a retransmission) until they have received a final response from the registrar for the previous one or the previous REGISTER request has timed out. The following Contact header parameters have a special meaning in REGISTER requests: action: The "action" parameter from RFC 2543 has been deprecated. UACs SHOULD NOT use the "action" parameter. expires: The "expires" parameter indicates how long the UA would like the binding to be valid. The value is a number indicating seconds. If this parameter is not provided, the value of the Expires header field is used instead. Implementations MAY treat values larger than 2**32-1 (4294967295 seconds or 136 years) as equivalent to 2**32-1. Various Authors [Page 50] Internet Draft SIP January 28, 2002 bob +----+ | UA | | | +----+ | |3)INVITE | carol@chicago.com chicago.com +--------+ V +---------+ 2)Store|Location|4)Query +-----+ |Registrar|=======>| Service|<=======|Proxy|sip.chicago.com +---------+ +--------+=======>+-----+ A 5)Resp | | | | | 1)REGISTER| | | | +----+ | | UA |<-------------------------------+ cube2214a| | 6)INVITE +----+ carol@cube2214a.chicago.com carol Figure 2: REGISTER example Various Authors [Page 51] Internet Draft SIP January 28, 2002 Malformed values should be treated as equivalent to 3600. 10.2.1 Adding Bindings The REGISTER request sent to a registrar includes contact addresses to which SIP requests for the address-of-record should be forwarded. The address-of-record is included in the To header field of the REGISTER request. The Contact header fields of the request typically contain SIP URIs that identify particular SIP endpoints (for example, "sip:carol@cube2214a.chicago.com"), but they MAY use any URI scheme. A SIP UA can choose to register telephone numbers (with the tel URL, [13]) or email addresses (with a mailto URL, [16]) as Contacts for an address-of-record. For example, Carol, with address-of-record "sip:carol@chicago.com", would register with the SIP registrar of the domain chicago.com. Her registrations would then be used by a proxy server in the chicago.com domain to route requests for Carol's address-of-record to her SIP endpoint. Once a client has established bindings at a registrar, it MAY send subsequent registrations containing new bindings or modifications to existing bindings as necessary. The 2xx response to the REGISTER request will contain, in Contact header fields, a complete list of bindings that have been registered for this address-of-record at this registrar. Registrations do not need to update all bindings. Typically, a UA only updates its own SIP URI as well as any non-SIP URIs. 10.2.1.1 Setting the Expiration Interval of Contact Addresses When a client sends a REGISTER request, it MAY suggest an expiration interval that indicates how long the client would like the registration to be valid. (As described in Section 10.3, the registrar selects the actual time interval based on its local policy.) There are two ways in which a client can suggest an expiration interval for a binding: through an Expires header field, or an "expires" Contact header parameter. The latter allows expiration intervals to be suggested on a per-binding basis when more than one binding is given in a single REGISTER request, whereas the former suggests an expiration interval for all Contact header fields that do not contain the "expires" parameter. Various Authors [Page 52] Internet Draft SIP January 28, 2002 If neither mechanism for expressing a suggested expiration time is present in a REGISTER, a default suggestion of one hour is assumed. 10.2.1.2 Preferences among Contact Addresses If more than one Contact is sent in a REGISTER request, the registering UA intends to associate all of the URIs given in these Contact headers with the address-of-record present in the To field. This list can be prioritized with the "q" parameter in the Contact header fields. The "q" parameter indicates a relative preference for the particular Contact header field compared to other bindings present in this REGISTER message or existing within the location service of the registrar. Section 16.5 describes how a proxy server uses this preference indication. 10.2.2 Removing Bindings Registrations are soft state and expire unless refreshed, but can also be explicitly removed. A client can attempt to influence the expiration interval selected by the registrar as described in Section 10.2.1. A user agent requests the immediate removal of a binding by specifying an expiration interval of "0" for that contact address in a REGISTER request. User agents SHOULD support this mechanism so that bindings can be removed before their expiration interval has passed. The REGISTER-specific Contact header field value of "*" applies to all registrations, but it MUST only be used when the Expires header field is present with a value of "0". Use of the "*" Contact header field value allows a registering user agent to remove all of its bindings without knowing their precise values. If no Contact header fields are present in a REGISTER request, the list of bindings is left unchanged. 10.2.3 Fetching Bindings A success response to any REGISTER request contains the complete list of existing bindings, regardless of whether the request contained a Contact header field or not. 10.2.4 Refreshing Bindings Each UA is responsible to refresh the bindings that it has previously established. A UA SHOULD NOT refresh bindings set up by other UAs. Various Authors [Page 53] Internet Draft SIP January 28, 2002 The 200 (OK) response from the registrar contains a list of Contact fields enumerating all current bindings. The UA compares each contact address to see if it created the contact address, using. comparison rules in Section 23.1.4. If so, it updates the expiration time interval according to the expires parameter or, if absent, the Expires field value. The UA then issues a REGISTER request for each of its bindings before the expiration interval has elapsed. It MAY combine several updates into one REGISTER request. A UA SHOULD use the same Call-ID for all registrations during a single boot cycle. Registration refreshes SHOULD be sent to the same network address as the original registration, unless redirected. 10.2.5 Setting the Internal Clock If the response for REGISTER request contains a Date header, the client MAY use this header field to learn the current time in order to set any internal clocks. 10.2.6 Discovering a Registrar UAs can use three ways to determine the address to send registrations to: by configuration, using the address-of-record and multicast. A UA can be configured, in ways beyond the scope of this specification, with a registrar address. If there is no configured registrar address, the UA SHOULD use the host part of the address-of-record as the Request-URI and address the request there, using the normal SIP server location mechanisms [8]. For example, the UA for the user "sip:carol@chicago.com" addresses the REGISTER request to "chicago.com". Finally, a UA can be configured to use multicast. Multicast registrations are addressed to the well-known "all SIP servers" multicast address "sip.mcast.net" (224.0.1.75 for IPv4). No well- known IPv6 multicast address has been allocated; such an allocation will be documented separately when needed. This request MUST be scoped to ensure it is not forwarded beyond the boundaries of the administrative system. This MAY be done with either TTL or administrative scopes (see [17]), depending on what is implemented in the network. SIP user agents MAY listen to that address and use it to become aware of the location of other local users (see [18]); however, they do not respond to the request. Multicast registration may be inappropriate in some environments, for example, if multiple businesses share the same local area network. Various Authors [Page 54] Internet Draft SIP January 28, 2002 10.2.7 Transmitting a Request Once the REGISTER method has been constructed, and the destination of the message identified, UACs should follow the procedures described in Section 8.1.2 to hand off the REGISTER to the transaction layer. If the transaction layer returns a timeout error because the REGISTER yielded no response, the UAC SHOULD wait some reasonable time interval before re-attempting a registration to the same registrar; no specific interval is mandated. 10.2.8 Error Responses If a UA receives a 423 (Registration Too Brief) response, it MAY retry the registration after making the expiration interval of all contact addresses in the REGISTER request equal to or greater than the expiration interval within the Min-Expires header of the 423 (Registration Too Brief) response. 10.3 Processing REGISTER Requests A registrar is a UAS that responds to REGISTER requests and maintains a list of bindings that are accessible to proxy servers within its administrative domain. A registrar handles requests according to Section 8.2 and Section 17.2, but it accepts only REGISTER requests. A registrar does not generate 6xx responses. If a registrar listens at a multicast interface, it MAY redirect multicast REGISTER requests to its own unicast interface with a 302 (Moved Temporarily) response. A REGISTER request MUST NOT contain Record-Route or Route header fields; registrars MUST ignore them if they appear. A registrar must know (e.g., through configuration) the set of domain(s) for which it maintains bindings. REGISTER requests MUST be processed by a registrar in the order that they are received. REGISTER requests MUST also be processed atomically, meaning that REGISTER requests are either processed completely or not at all. Each REGISTER message must be processed independently of any other registration or binding changes. When receiving a REGISTER request, a registrar follows these steps: 1. The registrar inspects the Request-URI to determine whether it has access to bindings for the domain identified in the Request-URI. If not and if the server also acts as a proxy server, the server SHOULD forward the request to the addressed domain, following the general behavior for proxying messages described in Section 16. Various Authors [Page 55] Internet Draft SIP January 28, 2002 2. To guarantee that the registrar supports any necessary extensions, the registrar processes Require header fields as described for UASs in Section 8.2.2. 3. A registrar SHOULD authenticate the UAC. Mechanisms for the authentication of SIP user agents are described in Section 20; registration behavior in no way overrides the generic authentication framework for SIP. If no authentication mechanism is available, the registrar MAY take the From address as the asserted identity of the originator of the request. 4. The registrar SHOULD determine if the authenticated user is authorized to modify registrations for this address-of- record. For example, a registrar might consult a authorization database that maps user names to a list of addresses-of-record for which this identity is authorized to modify bindings. If not, the registrar returns 403 (Forbidden) and skips the remaining steps. In architectures that support third-party registration, one entity may be responsible for updating the registrations associated with multiple addresses-of-record. 5. The registrar extracts the address-of-record from the To header field of request. If the address-of-record is not valid for the domain in the Request-URI, the registrar sends a 404 (Not Found) response and skips the remaining steps. The URI MUST then converted to a canonical form. To do that, all URI parameters are removed (including the user param), and any escaped characters are converted to their unescaped form. The result serves as an index into the list of bindings. 6. The registrar checks whether the request contains any Contact header fields. If not, it skips to the last step. Next, the registrar checks if there is one Contact field that contains the special value "*" and a Expires field. If the request has additional Contact fields or an expiration time other than zero, the request is invalid and the server returns 400 (Invalid Request) and skips the remaining steps. If not, the registrar checks whether the Call-ID agrees with the value stored for each binding. If not, it removes the binding. If it does agree, it only removes the binding if the CSeq in the request is higher than the value Various Authors [Page 56] Internet Draft SIP January 28, 2002 stored for that binding and leaves the binding as is otherwise. It then skips to the last step. 7. The registrar now processes each contact address in the Contact header field in turn. For each address, it determines the expiration interval as follows: - If the field value has an "expires" parameter, that value is used. - If there is no such parameter, but the request has an Expires header field, that value is used. - If there is neither, a locally-configured default value is used. The registrar MAY shorten the expiration interval. If and only if the expiration interval is greater than zero AND smaller than one hour AND less than a registrar-configured minimum, the registrar MAY reject the registration with a response of 423 (Registration Too Brief). This response MUST contain a Min-Expires header field that states the minimum expiration interval the registrar is willing to honor. It then skips the remaining steps. Allowing the registrar to set the registration interval protects it against excessively frequent registration refreshes while limiting the state that it needs to maintain and decreasing the likelihood of registrations going stale. The expiration interval of a registration is frequently used in the creation of services. An example is a follow-me service, where the user may only be available at a terminal for a brief period. Therefore, registrars should accept brief registrations; a request should only be rejected if the interval is so short that the refreshes would degrade registrar performance. For each address, it then searches the list of current bindings using the URI comparison rules. If the binding does not exist, it is tentatively added. If the binding does exist, the registrar checks the Call-ID value. If the existing binding has the same Call-ID value differs from the request, the binding is removed if the expiration time is zero and updated otherwise. If they are the same, the registrar compares the CSeq value. If the value is higher than that of the existing binding, it updates or removes Various Authors [Page 57] Internet Draft SIP January 28, 2002 the binding as above. If not, the update is aborted and the request fails. This algorithm ensures that out-of-order requests from the same UA are ignored. Each binding record records the Call-ID and CSeq values from the request. The binding updates are committed (i.e., made visible to the proxy) if and only if all binding updates and additions succeed. If any one of them fails, the request fails with 500 (Server Error) response and all tentative binding updates are removed. 8. The registrar returns a 200 (OK) response. The response MUST contain Contact header fields enumerating all current bindings. Each Contact value MUST feature an "expires" parameter indicating its expiration interval chosen by the registrar. The response SHOULD include a Date header field. 11 Querying for Capabilities The SIP method OPTIONS allows a UA to query another UA or a proxy server as to its capabilities. This allows a client to discover information about the methods, content types, extensions, codecs etc. supported without actually "ringing" the other party. For example, before a client inserts a Require header field into an INVITE listing an option that it is not certain the destination UAS supports, the client can query the destination UAS with an OPTIONS to see if this option is returned in a Supported header field. The target of the OPTIONS request is identified by the Request-URI, which could identify another User Agent or a SIP Server. If the OPTIONS is addressed to a proxy server, the Request-URI is set without a user part, similar to the way a Request-URI is set for a REGISTER request. Alternatively, a server receiving an OPTIONS request with a Max-Forwards header value of 0 MAY respond to the request regardless of the Request-URI. This behavior is common with HTTP/1.1. This behavior can be used as a "traceroute" functionality to check the capabilities of individual hop servers by sending a series of OPTIONS requests with incremented Max-Forwards values. As is the case for general UA behavior, the transaction layer can Various Authors [Page 58] Internet Draft SIP January 28, 2002 return a timeout error if the OPTIONS yields no response. This may indicate that the target is unreachable and hence unavailable. An OPTIONS request MAY be sent as part of an established dialog to query the peer on capabilities that may be utilized later in the dialog. 11.1 Construction of OPTIONS Request An OPTIONS request is constructed using the standard rules for a SIP request as discussed Section 8.1.1. A Contact header field MAY be present in an OPTIONS. An Accept header field SHOULD be included to indicate the type of message body the UAC wishes to receive in the response. Typically, this is set to a format that is used to describe the media capabilities of a UA, such as SDP (application/sdp). The response to an OPTIONS request is assumed to be scoped to the Request-URI in the original request. However, only when an OPTIONS is sent as part of an established dialog is it guaranteed that future requests will be received by the server which generated the OPTIONS response. Example OPTIONS request: OPTIONS sip:carol@chicago.com SIP/2.0 Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKhjhs8ass877 To: From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 63104 OPTIONS Contact: Accept: application/sdp Content-Length: 0 11.2 Processing of OPTIONS Request The response to an OPTIONS is constructed using the standard rules for a SIP response as discussed in Section 8.2.6. The response code chosen is the same that would have been chosen had the request been an INVITE. That is, a 200 (OK) would be returned if the UAS is ready to accept a call, a 486 (Busy Here) would be returned if the UAS is busy, etc. This allows an OPTIONS request to be used to determine the Various Authors [Page 59] Internet Draft SIP January 28, 2002 basic state of a UAS, which can be an indication of whether the UAC will accept an INVITE request. An OPTIONS request received within a dialog generates a 200 (OK) response which is identical to one constructed outside a dialog and does not have any impact on the dialog. This use of OPTIONS has limitations due the differences in proxy handling of OPTIONS and INVITE requests. While a forked INVITE can result in multiple 200 (OK) responses being returned, a forked OPTIONS will only result in a single 200 (OK) response, since it is treated by proxies using the non-INVITE handling. See Section 13.2.1 for the normative details. If the response to an OPTIONS is generated by a proxy server, the proxy returns a 200 (OK) listing the capabilities of the server. The response does not contain a message body. Allow, Accept, Accept-Encoding, Accept-Language, and Supported header fields SHOULD be present in a 200 (OK) response to an OPTIONS request. If the response is generated by a proxy, the Allow header field SHOULD be omitted as it is ambiguous since a proxy is method agnostic. Contact header fields MAY be present in a 200 (OK) response and have the same semantics as in a redirect. That is, they may list a set of alternative names and methods of reaching the user. A Warning header field MAY be present. A message body MAY be sent, the type of which is determined by the Accept header in the OPTIONS request (application/sdp if the Accept header was not present). If the types include one that can describe media capabilities, the UA SHOULD include a body in the response for that purpose. Details on construction of such a body in the case of application/sdp are described in [19]. Example OPTIONS response generated by a UAS (corresponding to the request in Section 11.1): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKhjhs8ass877 To: ;tag=93810874 From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710@100.1.3.3 CSeq: 63104 OPTIONS Contact: Contact: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Various Authors [Page 60] Internet Draft SIP January 28, 2002 Accept: application/sdp Accept-Encoding: gzip Accept-Language: en Supported: foo Content-Type: application/sdp Content-Length: 274 (SDP not shown) 12 Dialogs A key concept for a user agent is that of a dialog. A dialog represents a peer-to-peer SIP relationship between a two user agents that persists for some time. The dialog facilitates sequencing of messages between the user agents and proper routing of requests between both of them. The dialog represents a context in which to interpret SIP messages. Section 8 discussed method- independent UA processing for requests and responses outside of a dialog. This section discusses how those requests and responses are used to construct a dialog, and then how subsequent requests and responses are sent within a dialog. A dialog is identified at each UA with a dialog ID, which consists of a Call-ID value, a local URI and local tag (together called the local address), and a remote URI and remote tag (together called the remote address). The dialog ID at each UA involved in the dialog is not the same. Specifically, the local URI and local tag at one UA are identical to the remote URI and remote tag at the peer UA. The tags are opaque tokens that facilitate the generation of unique dialog IDs. A dialog ID is also associated with all responses and with any request that contains a tag in the To field. The rules for computing the dialog ID of a message depend on whether the entity is a UAC or UAS. For a UAC, the Call-ID value of the dialog ID is set to the Call-ID of the message, the remote address is set to the To field of the message, and the local address is set to the From field of the message (these rules apply to both requests and responses). As one would expect, for a UAS, the Call-ID value of the dialog ID is set to the Call-ID of the message, the remote address is set to the From field of the message, and the local address is set to the To field of the message. A dialog contains certain pieces of state needed for further message transmissions within the dialog. This state consists of the dialog ID, a local sequence number (used to order requests from the UA to Various Authors [Page 61] Internet Draft SIP January 28, 2002 its peer), a remote sequence number (used to order requests from its peer to the UA), and a route set, which is an ordered list of URIs. The route set is the set of servers that need to be traversed to send a request to the peer. A dialog can also be in the "early" state, which occurs when it is created with a provisional response, and then transition to the "confirmed" state when the final response comes. 12.1 Creation of a Dialog Dialogs are created through the generation of non-failure responses to requests with specific methods. Within this specification, only 2xx and 101-199 responses with a To tag to INVITE establish a dialog. A dialog established by a non-final response to a request is in the "early" state and it is called an early dialog. Extensions MAY define other means for creating dialogs. Section 13 gives more details that are specific to the INVITE method. Here, we describe the process for creation of dialog state that is not dependent on the method. A dialog is identified by a dialog ID. A dialog ID consists of three components, namely a call identifier component, a local address component and a remote address component. UAs MUST assign values to these components as described below. 12.1.1 UAS behavior When a UAS responds to a request with a response that establishes a dialog (such as a 2xx to INVITE), the UAS MUST copy all Record-Route headers from the request into the response (including the URIs, URI parameters, and any Record-Route header parameters, whether they are known or unknown to the UAS) and MUST maintain the order of those headers. The UAS MUST add a Contact header field to the response. The Contact header field contains an address where the UAS would like to be contacted for subsequent requests in the dialog (which includes the ACK for a 2xx response in the case of an INVITE). Generally, the host portion of this URI is the IP address or FQDN of the host. The URI provided in the Contact header field MUST be a SIP URI and have global scope (i.e., the same SIP URI can be used outside this dialog to contact the UAS). The same way, the scope of the SIP URI in the Contact header field of the INVITE is not limited to this dialog either. It can therefore be used to contact the UAC even outside this dialog. The UAS then constructs the state of the dialog. This state MUST be maintained for the duration of the dialog. First, the route set MUST be computed by following these steps: 1. The list of URIs in the Record-Route headers in the request, if present, are taken, including any URI Various Authors [Page 62] Internet Draft SIP January 28, 2002 parameters. 2. The URI in the Contact header from the request if present, is taken, including any URI parameters. The URI is appended to the bottom of the list of URIs from the previous step. Contact was not mandatory in RFC 2543. Thus, if the UAS is communicating with an older UAC, the UAC might not have inserted the Contact header field. 3. The resulting list of URIs is called the route set These rules clearly imply that a UA MUST be able to parse and process Record-Route header fields. This is a change from RFC 2543, where all record-route and route processing was optional for user agents. It is possible for the route set to be empty. This will occur if neither Record-Route headers nor a Contact header were present in the request. The UAS MUST also remember whether the bottom-most entry in the route set was constructed from a Contact header. This is effectively a boolean value, which we refer to as CONTACT_SET. From this value the UA can determine whether the bottom-most value can be updated from subsequent requests; if it was constructed from a Contact, it can be updated. The remote sequence number MUST be set to the value of the sequence number in the Cseq header of the request. The local sequence number MUST be empty. The call identifier component of the dialog ID MUST be set to the value of the Call-ID in the request. The local address component of the dialog ID MUST be set to the To field in the response to the request (which therefore includes the tag), and the remote address component of the dialog ID MUST be set to the From field in the request. A UAS MUST be prepared to receive a request without a tag in the From field, in which case the tag is considered to have a value of null. This is to maintain backwards compatibility with RFC 2543, which did not mandate From tags. 12.1.2 UAC behavior When a UAC receives a response that establishes a dialog, it constructs the state of the dialog. This state MUST be maintained for the duration of the dialog. First, the route set MUST be computed by following these steps: Various Authors [Page 63] Internet Draft SIP January 28, 2002 1. The list of URIs present in the Record-Route headers in the response are taken, if present, including all URI parameters, and their order is reversed. 2. The URI in the Contact header from the response, if present, is taken, including all URI parameters, and appended to the end of the list from the previous step. 3. The list of URIs resulting from the above two operations is referred to as the route set It is possible for the route set to be empty. This will occur if neither Record-Route headers nor a Contact header were present in the response. The UAC MUST also remember whether the bottom-most entry in the route set was constructed from a Contact header. This is effectively a boolean value, which we refer to as CONTACT_SET. From this value the UA can determine whether the bottom-most value can be updated from subsequent requests; if it was constructed from a Contact, it can be updated. The local sequence number MUST be set to the value of the sequence number in the Cseq header of the request. The remote sequence number MUST be empty (it is established when the UA sends a request within the dialog). The call identifier component of the dialog ID MUST be set to the value of the Call-ID in the request. The local address component of the dialog ID MUST be set to the From field in the request, and the remote address component of the dialog ID MUST be set to the To field of the response. A UAC MUST be prepared to receive a response without a tag in the To field, in which case the tag is considered to have a value of null. This is to maintain backwards compatibility with RFC 2543, which did not mandate To tags. 12.2 Requests within a Dialog Once a dialog has been established between two UAs, either of them MAY initiate new transactions as needed within the dialog. However, a dialog imposes some restrictions on the use of simultaneous transactions. A TU MUST NOT initiate a new regular transaction within a dialog while a regular transaction is in progress (in either direction) within that dialog. If there is a non-INVITE client or server transaction in progress the TU MUST wait until this transaction enters the completed or the terminated state to initiate the new transaction. Various Authors [Page 64] Internet Draft SIP January 28, 2002 OPEN ISSUE #113: Should we relax the constraint on non- overlapping regular transactions? A route refresh request sent within a dialog is defined as a request that can modify the route set of the dialog. For dialogs that have been established with an INVITE, the only route refresh request defined is re-INVITE (see Section 14). Other extensions may define different route refresh requests for dialogs established in other ways. Note that an ACK is NOT a route refresh request. 12.2.1 UAC Behavior 12.2.1.1 Generating the Request A request within a dialog is constructed by using many of the components of the state stored as part of the dialog. The To header field of the request MUST be set to the remote address, and the From header field MUST be set to the local address (both including tags, assuming the tags are not null). The Call-ID of the request MUST be set to the Call-ID of the dialog. Requests within a dialog MUST contain strictly monotonically increasing and contiguous CSeq sequence numbers (increasing-by-one) in each direction. Therefore, if the local sequence number is not empty, the value of the local sequence number MUST be incremented by one, and this value MUST placed into the Cseq header. If the local sequence number is empty, an initial value MUST be chosen using the guidelines of Section 8.1.1.5. The method field in the Cseq header MUST match the method of the request. With a length of 32 bits, a client could generate, within a single call, one request a second for about 136 years before needing to wrap around. The initial value of the sequence number is chosen so that subsequent requests within the same call will not wrap around. A non-zero initial value allows clients to use a time-based initial sequence number. A client could, for example, choose the 31 most significant bits of a 32-bit second clock as an initial sequence number. The Request-URI of requests is determined according to the following rules: The UAC takes the list of URI in the route set MUST be inserted into Various Authors [Page 65] Internet Draft SIP January 28, 2002 the Request-URI of the request, including all URI parameters. Any URI parameters not allowed in the Request-URI MUST then be stripped. Each of the remaining URIs (if any) from the route set , including all URI parameters, MUST be placed into a Route header field into the request, in order. A TU SHOULD follow the rules just mentioned to build the Request-URI of the request, regardless of whether the UA uses an outbound proxy server or not. However, in some instances, a UA may not be willing or capable of sending the request to the top element in the route set and therefore may not be able to follow those procedures. to use a loose-routing policy to send the request to its outbound proxy server (see section 8.1.3). This policy MUST include placing the topmost element in the route set as the first value in the message's Route header field as well as in the Request-URI. The loop-detection avoidance algorithm described in section 8.1.3 SHOULD be applied to the message before sending. A UAC SHOULD include a Contact header in any route refresh requests within a dialog, and unless there is a need to change it, the URI SHOULD be the same as used in previous requests within the dialog. As discussed in Section 12.2.2, a Contact header in a route refresh request updates the route set its address change during the duration of the dialog. However, requests that are not route refresh requests do not affect the route set for the dialog. Once the request has been constructed, the address of the server is computed and the request is sent, using the same procedures for requests outside of a dialog (Section 8.1.1). 12.2.1.2 Processing the Responses The UAC will receive responses to the request from the transaction layer. If the client transaction returns a timeout this is treated as a 408 (Request Timeout) response. The behavior of a UAC that receives a 3xx response for a request sent within a dialog is the same as if the request had been sent outside a dialog. This behavior is described in Section 13.2.2. Note, however, that when the UAC tries alternative locations, it still uses the route set for the dialog to build the Route header of the request. If a UAC has a route set for a dialog and receives a 2xx response to a route refresh it sent, the Contact header field of the response is Various Authors [Page 66] Internet Draft SIP January 28, 2002 examined. If not present, the route set remains unchanged. If the response had a Contact header field, and the boolean variable CONTACT_SET is false, the URI in the Contact header field in the response is added to the bottom of the route set , and CONTACT_SET is set to true. If the route refresh request response had a Contact header field, and CONTACT_SET is true, the URI in the Contact header field of the response to the route refresh request replaces the bottom value in the route set If a route refresh request is responded with a non-2xx final response the route set remains unchanged as if no route refresh request had been issued. If the response for the a request within a dialog is a 481 (Call/Transaction Does Not Exist) or a 408 (Request Timeout), the UAC SHOULD terminate the dialog. A UAC SHOULD also terminate a dialog if no response at all is received for the request (the client transaction would inform the TU about the timeout.) For INVITE initiated dialogs, terminating the dialog consists of sending a BYE. 12.2.2 UAS behavior Requests sent within a dialog, as any other requests, are atomic. If a particular request is accepted by the UAS, all the state changes associated with it are performed. If the request is rejected, none of the state changes is performed. Note that some requests such as INVITEs affect several pieces of state. The UAS will receive the request from the transaction layer. If the request has a tag in the To header field, the UAS core computes the dialog identifier corresponding to the request and compares it with existing dialogs. If there is a match, this is a mid-dialog request. In that case, the UAS applies the same processing rules for requests outside of a dialog, discussed in Section 8.2. If the request has a tag in the To header field, but the dialog identifier does not match any existing dialogs, the UAS may have crashed and restarted, or it may have received a request for a different (possibly failed) UAS (the UASs can construct the To tags so that a UAS can identify that the tag was for a UAS for which it is providing recovery). Another possibility is that the incoming request has been simply missrouted. Based on the To tag, the UAS MAY either accept or reject the request. Accepting the request for acceptable To tags provides robustness, so that dialogs can persist even through crashes. UAs wishing to support this capability must take into consideration some issues such as choosing monotonically increasing Various Authors [Page 67] Internet Draft SIP January 28, 2002 CSeq sequence numbers even across reboots, reconstructing the route set , and accepting out-of-range RTP timestamps and sequence numbers. If the UAS wishes to reject the request, because it does not wish to recreate the dialog, it MUST respond to the request with a 481 (Call/Transaction Does Not Exist) status code and pass that to the server transaction. Requests that do not change in any way the state of a dialog may be received within a dialog (for example, an OPTIONS request). They are processed as if they had been received outside the dialog. Requests within a dialog MAY contain Record-Route and Contact header fields. However, requests that are not route refresh requests do not update the route set for the dialog. This specification only defines one route refresh request: re-INVITE (see Section 14). Special rules apply when updated Record-Route or Contact header fields are received inside a route refresh request. If a UAS has a route set for a dialog and receives a route refresh for that dialog containing Record-Route header fields, it MUST copy those header fields into any 2xx response to that request. If the boolean variable CONTACT_SET is true, the Contact header field in the request (if present) replaces the last entry in the route set is false, the UAS MUST add the URI in the Contact header field in the route refresh request to the bottom of the route set , and then set CONTACT_SET to true. If the request did not contain a Contact header field, the route-set at the UAS remains unchanged. Route refresh requests only update the Contact of the route set and not the elements formed from Record-Route. Updating the latter would introduce severe backwards compatibility problems with RFC 2543-compliant systems. If the remote sequence number is empty, it MUST be set to the value of the sequence number in the Cseq header in the request. If the remote sequence number was not empty, but the sequence number of the request is lower than the remote sequence number, the request is out of order and MUST be rejected with a 500 (Server Internal Error) response. If the remote sequence number was not empty, and the sequence number of the request is greater than the remote sequence number, the request is in order. It is possible for the CSeq header to be higher than the remote sequence number by more than one. This is not an error condition, and a UAS SHOULD be prepared to receive and process requests with CSeq values more than one higher than the previous received request. The UAS MUST then set the remote sequence number to the value of the sequence number in the Cseq header in the Various Authors [Page 68] Internet Draft SIP January 28, 2002 request. If a proxy challenges a request generated by the UAC, the UAC has to resubmit the request with credentials. The resubmitted request will have a new Cseq number. The UAS will never see the first request, and thus, it will notice a gap in the Cseq number space. Such a gap does not represent any error condition. 12.3 Termination of a Dialog Dialogs can end in several different ways, depending on the method. When a dialog is established with INVITE, it is terminated with a BYE. No other means to terminate a dialog are described in this specification, but extensions can define other ways. 13 Initiating a Session 13.1 Overview When a user agent client desires to initiate a session (for example, audio, video, or a game), it formulates an INVITE request. The INVITE request asks a server to establish a session. This request is forwarded by proxies, eventually arriving at one or more UAS that can potentially accept the invitation. These UASs will frequently need to query the user about whether to accept the invitation. After some time, those UAS can accept the invitation (meaning the session is to be established) by sending a 2xx response. If the invitation is not accepted, a 3xx, 4xx, 5xx or 6xx response is sent, depending on the reason for the rejection. Before sending a final response, the UAS can also send a provisional response (1xx), either reliably or unreliably, to advise the UAC of progress in contacting the called user. After possibly receiving one or more provisional responses, the UA will get one or more 2xx responses or one non-2xx final response. Because of the protracted amount of time it can take to receive final responses to INVITE, the reliability mechanisms for INVITE transactions differ from those of other requests (like OPTIONS). Once it receives a final response, the UAC needs to send an ACK for every final response it receives. The procedure for sending this ACK depends on the type of response. For final responses between 300 and 699, the ACK processing is done in the transaction layer and follows one set of rules (See Section 17). For 2xx responses, the ACK is generated by the UAC core. A 2xx response to an INVITE establishes a session, and it also creates a dialog between the UA that issued the INVITE and the UA Various Authors [Page 69] Internet Draft SIP January 28, 2002 that generated the 2xx response. Therefore, when multiple 2xx responses are received from different remote UAs (because the INVITE forked), each 2xx establishes a different dialog. All these dialogs are part of the same call. This section provides details on the establishment of a session using INVITE. 13.2 Caller Processing 13.2.1 Creating the Initial INVITE Since the initial INVITE represents a request outside of a dialog, its construction follows the procedures of Section 8.1.1. Additional processing is required for the specific case of INVITE. An Allow header field (Section 24.5) SHOULD be present in the INVITE. It indicates what methods can be invoked within a dialog, on the UA sending the INVITE, for the duration of the dialog. For example, a UA capable of receiving INFO requests within a dialog [20] SHOULD include an Allow header listing the INFO method. A Supported header field (Section 24.39) SHOULD be present in the INVITE. It enumerates all the extensions understood by the UAC. An Accept (Section 24.1) header field MAY be present in the INVITE. It indicates which content-types are acceptable to the UA, in both the response received by it, and in any subsequent requests sent to it within dialogs established by the INVITE. The Accept header is especially useful for indicating support of various session description formats. The UA MAY add an Expires header field (Section 24.19) to limit the validity of the invitation. If the time indicated in the Expires header field is reached and no final answer for the INVITE has been received the UAC core SHOULD generate a CANCEL request for the original INVITE. A UAC MAY also find useful to add, among others, Subject (Section 24.38), Organization (Section 24.25) and User-Agent (Section 24.43) header fields. They all contain information related to the INVITE. The UAC MAY choose to add a message body to the INVITE. Section 8.1.1.10 deals with how to construct the header fields -- Content- Type among others -- needed to describe the message body. There are special rules for message bodies that contain a session description - their corresponding Content-Disposition is "session". Various Authors [Page 70] Internet Draft SIP January 28, 2002 SIP uses an offer/answer model where one UA sends a session description, called the offer, which contains a proposed description of the session. The offer indicates the desired communications means (audio, video, games), parameters of those means (such as codec types) and addresses for receiving media from the answerer. The other UA responds with another session description, called the answer, which indicates which communications means are accepted, the parameters which apply to those means, and addresses for receiving media from the offerer. The offer/answer model can be mapped into the INVITE transaction in two ways. The first, which is the most intuitive, is that the INVITE contains the offer, the 2xx response contains the answer, and no session description is provided in the ACK. In this model, the UAC is the offerer, and the UAS is the answerer. A second model is that the INVITE contains no session description, the 2xx response contains the offer, and the ACK contains the answer. In this model, the UAS is the offerer, and the UAC is the answerer. The second model is useful for gateways from H.323v1 to SIP, where the H.323 media characteristics are not known until the call is established. This is also useful for sessions that use third-party call control. As a result of these models, if the INVITE contains a session description, the ACK MUST NOT contain one. Conversely, if the caller chooses to omit the session description in the INVITE, the ACK MUST contain one (if a 2xx response is received). 2xx responses to an INVITE MUST always contain a session description. All user agents that support INVITE MUST support both models. The Session Description Protocol (SDP) [5] MUST be supported by all user agents as a means to describe sessions, and its usage for construction offers and answers MUST follow the procedures defined in [19]. The restrictions of the offer-answer model (session description only in the INVITE OR in the ACK, but not in both) just described only apply to bodies whose Content-Disposition header field is "session". Therefore, it is possible that both the INVITE and the ACK contain a body message (e.g., the INVITE carries a photo (Content-Disposition: render) and the ACK a session description (Content-Disposition: session) ). If the Content-Disposition header field is missing, bodies of Content-Type application/sdp imply the disposition "session", while other content types imply "render". Once the INVITE has been created, the UAC follows the procedures defined for sending requests outside of a dialog (Section 8). This results in the construction of a client transaction that will ultimately send the request and deliver responses to the UAC. Various Authors [Page 71] Internet Draft SIP January 28, 2002 13.2.2 Processing INVITE Responses Once the INVITE has been passed to the INVITE client transaction, the UAC waits for responses for the INVITE. Responses are matched to their corresponding INVITE because they have the same Call-ID, the same From header field, the same To header field, excluding the tag, and the same CSeq. Rules for comparisons of these headers are described in Section 24. If the INVITE client transaction returns a timeout rather than a response the TU acts as if a 408 (Request Timeout) response had been received. 13.2.2.1 1xx responses Zero, one or multiple provisional responses may arrive before one or more final responses are received. Provisional responses for an INVITE request can create "early dialogs". If a provisional response has a tag in the To field, and if the dialog ID of the response does not match an existing dialog, one is constructed using the procedures defined in Section 12.1.2. The early dialog will only be needed if the UAC needs to send a request to its peer within the dialog before the initial INVITE transaction completes. Header fields present in a provisional response are applicable as long as the dialog is in the early state (e.g., an Allow header field in a provisional response contains the methods that can be used in the dialog while this is in the early state). 13.2.2.2 3xx responses A 3xx response may contain a Contact header field providing new addresses where the callee might be reachable. Depending on the status code of the 3xx response (see Section 25.3) the UAC MAY choose to try those new addresses. 13.2.2.3 4xx, 5xx and 6xx responses A single non-2xx final response may be received for the INVITE. 4xx, 5xx and 6xx responses may contain a Contact header field indicating the location where additional information about the error can be found. All early dialogs are considered terminated upon reception of the non-2xx final response. After having received the non-2xx final response the UAC core considers the INVITE transaction completed. The INVITE client transaction handles generation of ACKs for the response (see Section Various Authors [Page 72] Internet Draft SIP January 28, 2002 17). 13.2.2.4 2xx responses Multiple 2xx responses may arrive at the UAC for a single INVITE request due to a forking proxy. Each response is distinguished by the tag parameter in the To header field, and each represents a distinct dialog, with a distinct dialog identifier. If the dialog identifier in the 2xx response matches the dialog identifier of an existing dialog, the dialog MUST be transitioned to the "confirmed" state, and the route set for the dialog MUST be recomputed based on the 2xx response using the procedures of Section 12.1.2. Otherwise, a new dialog in the "confirmed" state is constructed in the same fashion. The route set only is recomputed for backwards compatibility. RFC 2543 did not mandate mirroring of Record-Route headers in a 1xx, only 2xx. However, we cannot update the entire state of the dialog, since mid-dialog requests may have been sent within the early call leg, modifying the sequence numbers, for example. The UAC core MUST generate an ACK request for each 2xx received from the transaction layer. The header fields of the ACK are constructed in the same way as for any request sent within a dialog (see Section 12) with the exception of the CSeq and the header fields related to authentication. The sequence number of the CSeq header field MUST be the same as the INVITE being acknowledged, but the CSeq method MUST be ACK. The ACK MUST contain the same credentials as the INVITE. If the INVITE did not contain an offer, the 2xx will contain one, and therefore the ACK MUST carry an answer in its body. If the offer in the 2xx response is not acceptable the UAC core MUST generate a valid answer in the ACK and then send a BYE immediately. Once the ACK has been constructed, the procedures of [8] are used to determine the destination address, port and transport. However, the request is passed to the transport layer directly for transmission, rather than a client transaction. This is because the UAC core handles retransmissions of the ACK, not the transaction layer. The ACK MUST be passed to the client transport every time a retransmission of the 2xx final response that triggered the ACK arrives. The UAC core considers the INVITE transaction completed 64*T1 seconds after the reception of the first 2xx response. At this point all the early dialogs that have not transitioned to established dialogs are Various Authors [Page 73] Internet Draft SIP January 28, 2002 terminated. Once the INVITE transaction is considered completed by the UAC core, no more new 2xx responses are expected to arrive. If, after acknowledging any 2xx response to an INVITE, the caller does not want to continue with that dialog, then the caller MUST terminate the dialog by sending a BYE request as described in Section 15. 13.3 Callee Processing 13.3.1 Processing of the INVITE The UAS core will receive INVITE requests from the transaction layer. It first performs the request processing procedures of Section 8.2, which are applied for both requests inside and outside of a dialog. Assuming these processing states complete without generating a response, the UAS core performs the additional processing steps: 1. If the request is an INVITE that contains an Expires header field the UAS core inspects this header field. If the INVITE has already expired a 487 (Request Terminated) response SHOULD be generated. In any case, if the INVITE expires before the UAS has generated a final response a 487 (Request Terminated) response SHOULD be generated. 2. If the request is a mid-dialog request, the method- independent processing described in Section 12.2.2 is first applied. It might also modify the session; Section 14 provides details. 3. If the request has a tag in the To header field but the dialog identifier does not match any of the existing dialogs, the UAS may have crashed and restarted, or may have received a request for a different (possibly failed) UAS. Section 12.2.2 provides guidelines to achieve a robust behaviour under such a situation. Processing from here forward assumes that the INVITE is outside of a dialog, and is thus for the purposes of establishing a new session. The INVITE may contain a session description, in which case the UAS is being presented with an offer for that session. It is possible that the user is already a participant in that session, even though the INVITE is outside of a dialog. This can happen when a user is invited to the same multicast conference by multiple other participants. If desired, the UAS MAY use identifiers within the session description to detect this duplication. For example, SDP Various Authors [Page 74] Internet Draft SIP January 28, 2002 contains a session id and version number in the origin (o) field. If the user is already a member of the session, and the session parameters contained in the session description have not changed, the UAS MAY silently accept the INVITE (that is, send a 2xx response without prompting the user). The INVITE may not contain a session description at all, in which case the UAS is being asked to participate in a session, but the UAC has asked that the UAS provide the offer of the session. The callee can indicate progress, accept, redirect, or reject the invitation. In all of these cases, it formulates a response using the procedures described in Section 8.2.6. 13.3.1.1 Progress The UAS may not be able to answer the invitation immediately, and might choose to indicate some kind of progress to the caller (for example, an indication that a phone is ringing). This is accomplished with a provisional response between 101 and 199. These provisional responses establish early dialogs and therefore follow the procedures of Section 12.1.1 in addition to those of Section 8.2.6. A UAS MAY send as many provisional responses as it likes. Each of these MUST indicate the same dialog ID. However, these will not be delivered reliably unless reliable provisional responses are used. If the UAS will require an extended period of time to answer the INVITE, it will need to ask for an "extension" in order to prevent proxies from cancelling the transaction. A proxy has the option of canceling a transaction when there is a gap of 3 minutes between messages in a transaction. To prevent cancellation, the UAS MUST send a non-100 provisional response at least that often. This response SHOULD be sent reliably, if supported by the UAC. If not, the UAS SHOULD send provisional responses every minute, to handle the possibility of lost provisional responses. An INVITE transaction can go on for extended durations when the user is placed on hold, or when interworking with PSTN systems which allow communications to take place without answering the call. The latter is common in Interactive Voice Response (IVR) systems. 13.3.1.2 The INVITE is redirected If the UAS decides to redirect the call, a 3xx response is sent. A 300 (Multiple Choices), 301 (Moved Permanently) or 302 (Moved Temporarily) response SHOULD contain a Contact header field Various Authors [Page 75] Internet Draft SIP January 28, 2002 containing URIs of new addresses to be tried. The response is passed to the INVITE server transaction, which will deal with its retransmissions. 13.3.1.3 The INVITE is rejected A common scenario occurs when the callee is currently not willing or able to take additional calls at this end system. A 486 (Busy Here) SHOULD be returned in such scenario. If the UAS knows that no other end system will be able to accept this call a 600 (Busy Everywhere) response SHOULD be sent instead. However, it is unlikely that a UAS will be able to know this in general, and thus this response will not usually be used. The response is passed to the INVITE server transaction, which will deal with its retransmissions. A UAS rejecting an offer contained in an INVITE SHOULD return a 488 (Not Acceptable Here) response. Such a response SHOULD include a Warning header field explaining why the offer was rejected. 13.3.1.4 The INVITE is accepted The UAS core generates a 2xx response. This response establishes a dialog, and therefore follows the procedures of Section 12.1.1 in addition to those of Section 8.2.6. A 2xx response to an INVITE SHOULD contain the Allow header field and the Supported header field, and MAY contain the Accept header field. Including these header fields allows the UAC to determine the features and extensions supported by the UAS for the duration of the call, without probing. If the INVITE request contained an offer, the 2xx MUST contain an answer. If the INVITE did not contain an offer, the 2xx MUST contain an offer. Once the response has been constructed it is passed to the INVITE server transaction. Note, however, that the INVITE server transaction will be destroyed as soon as it receives this final response. Therefore, it is necessary to pass periodically the response to the transport until the ACK arrives. The 2xx response is passed to the transport with an interval that starts at T1 seconds and doubles for each retransmission until it reaches T2 seconds (T1 and T2 are defined in Section 17). Response retransmissions cease when an ACK request is received with the same dialog ID as the response. This is independent of whatever transport protocols are used to send the response. Various Authors [Page 76] Internet Draft SIP January 28, 2002 Since 2xx is retransmitted end-to-end, there may be hops between UAS and UAC which are UDP. To ensure reliable delivery across these hops, the response is retransmitted periodically even if the transport at the UAS is reliable. If the server retransmits the 2xx response for 64*T1 seconds without receiving an ACK, it considers the dialog completed, the session terminated, and therefore it SHOULD send a BYE. 14 Modifying an Existing Session A successful INVITE request (see Section 13) establishes both a dialog between two user agents and a session (using the offer/answer model). Section 12 explains how to modify an existing dialog using a route refresh request (e.g., changing the route set of the dialog). This section describes how to modify the actual session. This modification can involve changing addresses or ports, adding a media stream, deleting a media stream, and so on. This is accomplished by sending a new INVITE request within the same dialog that established the session. An INVITE request sent within an existing dialog is known as a re-INVITE. Note that a single re-INVITE can modify at the same time the dialog and the parameters of the session. Either the caller or callee can modify an existing session. The behaviour of a UA on detection of media failure is a matter of local policy. However, automated generation of re-INVITE or BYE is NOT RECOMMENDED to avoid flooding the network with traffic when there is congestion. In any case, if these messages are sent automatically, they SHOULD be sent after some randomized interval. Note that the paragraph above refers to automatically generated BYEs and re-INVITEs. If the user hangs up upon media failure the UA would send a BYE request as usual. 14.1 UAC Behavior The same offer-answer model that applies to session descriptions in INVITEs (Section 13.2.1) applies to re-INVITEs. As a result, a UAC that wants to add a media stream, for example, will create a new offer that contains this media stream, and send that in an INVITE request to its peer. It is important to note that the full description of the session, not just the change, is sent. This maintains the idempotency of SIP, supports stateless session processing in various elements, and supports failover and recovery Various Authors [Page 77] Internet Draft SIP January 28, 2002 capabilities. Of course, a UAC MAY send a re-INVITE with no session description, in which case the response to the re-INVITE will contain the offer. If the session description format has the capability for version numbers, the offerer SHOULD indicate that the version of the session description has changed. The To, From, Call-ID, CSeq, and Request-URI of a re-INVITE are set following the same rules as for regular requests within an existing dialog, described in Section 12. A UAC MAY choose not to add Alert-Info header fields or bodies with Content-Disposition "alert" to re-INVITEs because UASs do not typically alert the user upon reception of a re-INVITE. Note that, as opposed to initial INVITEs (see Section 13), re-INVITEs contain tags in the To header field and are sent using the route set for the dialog. Therefore, a single final (2xx or non-2xx) response is received for re-INVITEs. Note that a UAC MUST NOT initiate a new INVITE transaction within a dialog while another transaction (INVITE or non-INVITE) is in progress in either direction. 1. If there is an ongoing INVITE client transaction the TU MUST wait until the transaction reaches the completed or terminated state before initiating the new INVITE. 2. If there is an ongoing INVITE server transaction the TU MUST wait until the transaction reaches the confirmed or terminated state before initiating the new INVITE. 3. If there is an ongoing non-INVITE client or server transaction the TU MUST wait until the transaction reaches the completed or terminated state before initiating the new INVITE. However, a UA MAY initiate a regular transaction while an INVITE transaction is in progress. If a re-INVITE is responded with a non-2xx final response the session parameters MUST remain unchanged, as if no re-INVITE had been issued. Note that, as stated in Section 12.2.1.2, if the non-2xx final response is a 481 (Call/Transaction Does Not Exist) or a 408 (Request Timeout) or no response at all is received for the re-INVITE (a timeout is returned by the INVITE client transaction) the UAC will terminate the dialog. Various Authors [Page 78] Internet Draft SIP January 28, 2002 The rules for transmitting a re-INVITE and for generating an ACK for a 2xx response to re-INVITE are the same as for an INVITE (Section 13.2.1). 14.2 UAS Behavior Section 13.3.1 describes the steps to follow in order to distinguish incoming re-INVITEs from incoming initial INVITEs. This Section describes the procedures to follow upon reception of a re-INVITE for an existing dialog. A UAS that receives a second INVITE before it sent the final response to a first INVITE with a lower CSeq sequence number on the same dialog MUST return a 500 (Server Internal Error) response to the second INVITE and MUST include a Retry-After header field with a randomly chosen value of between 0 and 10 seconds. A UAS that receives an INVITE on a dialog while an INVITE it had sent on that dialog is in progress MUST return a 491 (Request Pending) response to the received INVITE and MUST include a Retry-After header field with a value chosen as follows: 1. If the UAS is the owner of the Call-ID of the dialog ID the Retry-After header field has a randomly chosen value of between 2.1 and 4 seconds in units of 10 ms. 2. If the UAS is not the owner of the Call-ID of the dialog ID the Retry-After header field has a randomly chosen value of between 0 and 2 seconds in units of 10 ms. If a user agent receives a re-INVITE for an existing dialog it MUST check any version identifiers in the session description or, if there are no version identifiers, the content of the session description to see if it has changed. If the session description has changed, the user agent server MUST adjust the session parameters accordingly, possibly after asking the user for confirmation. Versioning of the session description can be used to accommodate the capabilities of new arrivals to a conference, add or delete media or change from a unicast to a multicast conference. If the new session description is not acceptable the UAS can reject it by returning a 488 (Not Acceptable Here) response for the re-INVITE. This response SHOULD include a Warning header field. If a UAS generates a 2xx response and never receives an ACK, it SHOULD generate a BYE to terminate the dialog. Various Authors [Page 79] Internet Draft SIP January 28, 2002 A UAS MAY choose not to generate 180 (Ringing) responses for a re- INVITE because UACs do not typically render this information to the user. For the same reason UASs MAY choose not to use Alert-Info header fields or bodies with Content-Disposition "alert" in responses to a re-INVITE either. A UAS providing an offer in a 2xx (because the INVITE did not contain an offer) MUST offer the same session description as last provided to the peer, with the exception of being able to change the IP address/port if so desired. Under error conditions (e.g., the UAS has crashed and restarted) the session description in the 2xx response for an empty re-INVITE may be different than the one in use at that moment. If the new session description is not acceptable for the UAC it SHOULD then send a BYE (after ACKing the 2xx response). 15 Terminating a Session This section describes the procedures to be followed in order to terminate a SIP dialog. For two-party sessions that are otherwise unbound in time the termination of the dialog implies the termination of the session. Other types of sessions such as multicast sessions are not terminated when a participant terminates the SIP dialog that he used to join the session. However, the SIP dialog SHOULD be terminated even though its termination does not imply the termination of the session. A UA joining a multicast session MAY terminate the SIP dialog immediately after the INVITE transaction used to join the session has completed. Either the caller or callee may terminate a dialog for any reason. A caller terminates a dialog either with BYE of CANCEL depending on the state of the dialog. A callee uses BYE to terminate a confirmed dialog. If the callee wants to terminate an early dialog it just returns a non-2xx final response for the INVITE. Sections 13 and 12 document some cases where dialog termination is normative behavior. As a general rule, if a UA decides that the dialog is to be terminated, it MUST follow the procedures here to initiate signaling action to convey that. When a UAC sends an INVITE request to create a session, if a 1xx response with a tag in the To field is received, an early dialog is created. When a 2xx response is received, the dialog becomes confirmed. For a confirmed dialog, if the UAC desires to terminate Various Authors [Page 80] Internet Draft SIP January 28, 2002 the session, the UAC SHOULD follow the procedures described in Section 15.1.1 to terminate the session. If the callee for a new session wishes to terminate the dialog, it uses the procedures of Section 15.1.1, but MUST NOT do so until it has received an ACK or until the server transaction times out. This does not mean a user can't hang up right away; it just means that the software in their phone needs to maintain state for a short while in order to properly clean up. If the UAC desires to end the session before a confirmed dialog has been created, it SHOULD send a CANCEL for the INVITE request that requested establishment of the session that is to be terminated. The UAC constructs and sends the CANCEL following the procedures described in Section 9. This CANCEL will normally result in a 487 (Request Terminated) response to be returned to the INVITE, indicating successful cancellation. However, it is possible that the CANCEL and a 2xx response to the INVITE "pass on the wire". In this case, the UAC will receive a 2xx to the INVITE. It SHOULD then terminate the call by following the procedures described in Section 15.1.1. A UAC can terminate a specific early dialog by following the procedures described in Section 15.1.1. This would only terminate one particular early dialog. 15.1 Terminating a Dialog with a BYE Request 15.1.1 UAC Behavior A user agent client uses BYE request, sent within a dialog, to indicate to the server that it wishes to terminate the session. This will also terminate the dialog. A BYE request MAY be issued by either caller or callee. A BYE request SHOULD NOT be sent before the creation of a dialog (either early or confirmed). In that case the UAC SHOULD follow the procedures described in Section 9 instead. Proxies ensure that a CANCEL request is routed in the same way as the INVITE was. However, a proxy performing load balancing may route a BYE without a Route header field in a different way than the INVITE, since both requests have different CSeq sequence numbers. The To, From, Call-ID, CSeq, and Request-URI of a BYE are set following the same rules as for regular requests sent within a dialog, described in Section 12. Once the BYE is constructed, it creates a new non-INVITE client Various Authors [Page 81] Internet Draft SIP January 28, 2002 transaction, and passes it the BYE request. The user agent SHOULD stop sending media as soon as the BYE request is passed to the client transaction. If the response for the BYE is a a 481 (Call/Transaction Does Not Exist) or a 408 (Request Timeout) or no response at all is received for the BYE (a timeout is returned by the client transaction) the UAC considers the dialog down anyway. 15.1.2 UAS Behavior A UAS first processes the BYE request according to the general UAS processing described in Section 8.2. A UAS core receiving a BYE request checks to see if it matches an existing dialog. If the BYE does not match an existing dialog, the UAS core SHOULD generate a 481 (Call/Transaction Does Not Exist) response and pass that to the server transaction. This rule means that a BYE sent without tags by a UAC will be rejected. This is a change from RFC 2543, which allowed BYE without tags. A UAS core receiving a BYE request for an existing dialog MUST follow the procedures of Section 12.2.2 to process the request. Once done, the UAS MUST cease transmitting media streams for the session being terminated. The UAS core MUST generate a 2xx response to the BYE, and MUST pass that to the server transaction for transmission. The UAS MUST still respond to any pending requests received for that dialog, (which can only be an INVITE). It is RECOMMENDED that a 487 (Request Terminated) response is generated to those pending requests. 16 Proxy Behavior 16.1 Overview SIP proxies are elements that route SIP requests to user agent servers and SIP responses to user agent clients. A request may traverse several proxies on its way to a UAS. Each will make routing decisions, modifying the request before forwarding it to the next element. Responses will route through the same set of proxies traversed by the request in the reverse order. Being a proxy is a logical role for a SIP element. When a request arrives, an element that can play the role of a proxy must first decide if it needs to respond to the request on its own. For instance, the request could be malformed or the element may need credentials from the client before acting as a proxy. The element MAY respond with any appropriate error code. When responding directly to Various Authors [Page 82] Internet Draft SIP January 28, 2002 a request, the element is playing the role of a UAS and MUST behave as described in Section 8.2. A proxy can operate in either a stateful or stateless mode for each new request. When stateless, a proxy acts as a simple forwarding element. It forwards each request downstream to a single element determined by making a routing decision based on the request. It simply forwards every response it receives upstream. A stateless proxy discards information about a message once it has been forwarded. On the other hand, a stateful proxy remembers information (specifically, transaction state) about each incoming request and any requests it sends as a result of processing the incoming request. It uses this information to affect the processing of future messages associated with that request. A stateful proxy MAY chose to "fork" a request, routing it to multiple destinations. Any request that is forwarded to more than one location MUST be handled statefully. Any request processed using TCP (or any other mechanism that is inherently stateful), MUST be handled statefully. A stateful proxy MAY transition to stateless operation at any time during the processing of a request, so long as it did not do anything that would otherwise prevent it from being stateless initially (forking, for example, or generation of a 100 response). When performing such a transition, all state is simply discarded. The proxy SHOULD NOT send a CANCEL. Much of the processing involved when acting statelessly or statefully for a request is identical. The next several subsections are written from the point of view of a stateful proxy. The last section calls out those places where a stateless proxy behaves differently. 16.2 Stateful Proxy When stateful, a proxy is purely a SIP transaction processing engine. Its behavior is modeled here in terms of the Server and Client Transactions defined in Section 17. A stateful proxy has a server transaction associated with one or more client transactions by a higher layer proxy processing component (see figure 3), known as a proxy core. An incoming request is processed by a server transaction. Requests from the server transaction are passed to a proxy core. The proxy core determines where to route the request, choosing one or more next-hop locations. An outgoing request for each next-hop location is processed by its own associated client transaction. The proxy core collects the responses from the client transactions and uses them to send responses to the server transaction. Various Authors [Page 83] Internet Draft SIP January 28, 2002 A stateful proxy creates a new server transaction for each new request received. Any retransmissions of the request will then be handled by that server transaction per Section 17. This is a model of proxy behavior, not of software. An implementation is free to take any approach that replicates the external behavior this model defines. For all new requests, including any with unknown methods, an element intending to proxy the request MUST: 1. Validate the request (Section 16.3) .IP 2. Make a routing decision (Section 16.4) .IP 3. Forward the request to each chosen destination (Section 16.5) .IP 4. Process all responses (Section 16.6) 16.3 Request Validation Before an element can proxy a request, it MUST verify the message's validity. A valid message must pass the following checks: 1. Reasonable Syntax 2. Max-Forwards 3. (Optional) Loop Detection 4. Proxy-Require 5. Proxy-Authorization If any of these checks fail, the element MUST behave as a user agent server (see Section 8.2) and respond with an error code. Notice that a proxy is not required to detect merged requests and MUST NOT treat merged requests as an error condition. The endpoints receiving the requests will resolve the merge as described in Section 8.2.2.2. 1. Reasonable Syntax check The request MUST be well-formed enough to be handled with a server transaction. Any components involved in the remainder of these Request Validation steps or the Request Processing section MUST be well-formed. Any other components, well-formed or not, SHOULD be ignored and remain unchanged when the message is forwarded. For Various Authors [Page 84] Internet Draft SIP January 28, 2002 +------------------------------+ | | +---+ | | | T| | | | r| | | |C a| | | |l n| | | |i s| | | |e a| | | |n c| | | |t t| | | | i| | | | o| | | | n| | | +---+ +---+ | | +---+ | T| | | | T| | r| | | | r| |S a| | | |C a| |e n| | Proxy | |l n| |r s| | "Higher" Layer | |i s| |v a| | | |e a| |e c| | | |n c| |r t| | | |t t| | i| | | | i| | o| | | | o| | n| | | | n| +---+ | | +---+ | | +---+ | | | T| | | | r| | | |C a| | | |l n| | | |i s| | | |e a| | | |n c| | | |t t| | | | i| | | | o| | | | n| | | +---+ +------------------------------+ Figure 3: Stateful Proxy Model instance, an element SHOULD NOT reject a request because of a malformed Date header field. Likewise, a proxy SHOULD NOT remove a malformed Date header before forwarding a Various Authors [Page 85] Internet Draft SIP January 28, 2002 This protocol is designed to be extended. Future extensions may define new methods and header fields at any time. An element MUST NOT refuse to proxy a request because it contains a method or header field it does not know about. 2. Max-Forwards check The Max-Forwards header (Section 24.22) is used to limit the number of elements a SIP request can traverse. If the request does not contain a Max-Forwards header field, this check is passed. If the request contains a Max-Forwards header field with a field value greater than zero, the check is passed. If the request contains a Max-Forwards header field with a field value of zero (0), the element MUST NOT forward the request. If the request was for OPTIONS, the element MAY act as the final recipient and respond per Section 11. Otherwise, the element MUST return a 483 (Too many hops) response. 3. Optional Loop Detection check An element MAY check for forwarding loops before forwarding a request. If the request contains a Via header field value with A sent-by value that equals a value placed into previous requests by the proxy, the request has been forwarded by this element before. The request has either looped or is legitimately spiraling through the element. To determine if the request has looped, the element MAY perform the branch parameter calculation described in Step 3 of Section 16.5 on this message and compare it to the parameter received in that Via field value. If the parameters match, the request has looped. If they differ, the request is spiraling, and processing continues. If a loop is detected, the element MAY return a 482 (Loop Detected) response. In earlier versions of this memo, loop detection was REQUIRED. This requirement has been relaxed in favor of the Max-Forwards mechanism. Various Authors [Page 86] Internet Draft SIP January 28, 2002 4. Proxy-Require check Future extensions to this protocol may introduce features that require special handling by proxies. Endpoints will include a Proxy-Require header in requests that use these features, telling the proxy it should not process the request unless the feature is understood. If the request contains a Proxy-Require header (Section 24.29) with one or more option-tags this element does not understand, the element MUST return a 420 (Bad Extension) response. The response MUST include an Unsupported (Section 24.42) header field listing those option-tags the element did not understand. 5. Proxy-Authorization check If an element requires credentials before forwarding a request, the request MUST be inspected as described in Section 20.3. That section also defines what the element must do if the inspection fails. 16.4 Making a Routing Decision At this point, the proxy must decide where to forward the request. This can be modeled as computing a set of destinations for the request. This set will either be predetermined by the contents of the request or will be obtained from an abstract location service. Each destination is represented as a URI and an optional IP address, port and transport. This combination is referred to as a "next-hop location". First, the proxy core checks the received request for Route headers. If any Route header fields are present in the request, the proxy MUST choose a single next-hop location to place in the destination set. The proxy SHOULD choose to use a strict-routing policy, placing the URI (including all of its parameters) from the topmost Route header field as the only next hop URI in the destination set, with no IP address, port and transport set for that next hop. The proxy MAY choose to use a loose-routing policy, selecting a URI, address, port and transport based on that policy. A loose-routing policy MAY use any information in or about the request in determining where to route it. Restrictions on the a loose-routing proxy's policy are discussed in Section 8.1.3. Once the single next-hop location is placed into the destination set, the set is complete, and the proxy MUST proceed to the Request Processing of Section 16.5. Various Authors [Page 87] Internet Draft SIP January 28, 2002 The Route mechanism is used to affect the path a request takes through SIP elements. A strict-routing policy results in behaviour much like strict IP source routing. Loose-routing policies will result in the specified URIs being reached, possibly visiting additional elements in the process. A UAC will insert Route header fields (see Section 12), based on information provided by proxies through Record-Route header fields or by policy obtained through configuration. (see Step 6 of Section 16.5). Assuming there were no Route headers in the received request, the proxy checks the Request-URI of the received request. If the Request-URI has a URI whose scheme is not understood by the proxy, the proxy SHOULD reject the request with a 416 (Unsupported URI Scheme) response. If the Request-URI contains an maddr parameter, the proxy MUST check to see if its value is in the set of addresses or domains the proxy is configured to be responsible for. If the Request-URI has an maddr parameter with a value the proxy is responsible for, and the request was received using the port and transport indicated (explicitly or by default) in the Request-URI, the proxy MUST strip the maddr and any non-default port or transport parameter and continue processing as if those values had not been present in the request. Otherwise, if the Request-URI contains an maddr parameter, the Request-URI MUST be placed into the destination set as the only next hop URI, with no IP address, port and transport set for that next hop, and the proxy MUST proceed to Section 16.5. A request may arrive with an maddr matching the proxy, but on a port or transport different from that indicated in the URI. Such a request needs to be forwarded to the proxy using the indicated port and transport. If the domain of the Request-URI indicates a domain this element is not responsible for, it SHOULD set the next hop URI to the Request- URI, and leave the IP address, port and transport of the next hop empty. That next hop MUST be placed into the destination set as the only next hop, and the element MUST proceed to the task of Request Processing (Section 16.5. There are many circumstances in which a proxy might receive a request for a domain it is not responsible for. A firewall proxy handling outgoing calls (the way HTTP proxies handle outgoing requests) is an example of where this is likely to occur. If the destination set for the request has not been predetermined as described above, this implies that the element is responsible for the Various Authors [Page 88] Internet Draft SIP January 28, 2002 domain in the Request-URI, and the element MAY use whatever mechanism it desires to determine where to send the request. Any of these mechanisms can be modeled as accessing an abstract Location Service. This may consist of obtaining information from a location service created by a SIP Registrar, reading a database, consulting a presence server, utilizing other protocols, or simply performing an algorithmic substitution on the Request-URI. When accessing the location service constructed by the registrar, the Request-URI MUST first be canonicalized as described in Section 10.3 before being used as an index. The output of these mechanisms is used to construct the destination set. If the Request-URI does not provide sufficient information for the proxy to determine the destination set, it SHOULD return a 485 (Ambiguous) response. This response SHOULD contain a Contact header field containing URIs of new addresses to be tried. For example, an INVITE to sip:John.Smith@company.com may be ambiguous at a proxy whose location service has multiple John Smiths listed. See Section 25.4.23 for details. Any information in or about the request or the current environment of the element MAY be used in the construction of the destination set. For instance, different sets may be constructed depending on contents or the presence of header fields and bodies, the time of day of the request's arrival, the interface on which the request arrived, failure of previous requests, or even the element's current level of utilization. As potential destinations are located through these services, their next hops are added to the destination set. Next-hop locations may only be placed in the destination set once. If a next-hop location is already present in the set (based on the definition of equality for the URI type and equality of the optional parameters), it MUST NOT be added again. If the recieved request contained no Route headers, a proxy MAY continue to add destinations to the set after beginning Request Processing. It MAY use any information obtained during that processing to determine new locations. For instance, a proxy may choose to incorporate contacts obtained in a redirect response (3xx class) into the destination set. If a proxy uses a dynamic source of information while building the destination set (for instance, if it consults a SIP Registrar), it SHOULD monitor that source for the duration of processing the request. New locations SHOULD be added to the destination set as they become available. As above, any given URI MUST NOT be added to the set more than once. Various Authors [Page 89] Internet Draft SIP January 28, 2002 Allowing a URI to be added to the set only once reduces unnecessary network traffic, and in the case of incorporating contacts from redirect requests prevents infinite recursion. An example trivial location service is achieved by configuring an element with a default outbound destination. All requests are forwarded to this location. The Request-URI of the request is placed in the destination set with the optional next-hop IP address, port and transport parameters set to the default outbound destination. The destination set is complete, containing only this URI, and the element proceeds to the task of Request Processing. If the Request-URI indicates a resource at this proxy that does not exist, the proxy MUST return a 404 (Not Found) response. If the destination set remains empty after applying all of the above, the proxy MUST return an error response, which SHOULD be the 480 (Temporarily Unavailable) response. 16.5 Request Processing As soon as the destination set is non-empty, a proxy MAY begin forwarding the request. A stateful proxy MAY process the set in any order. It MAY process multiple destinations serially, allowing each client transaction to complete before starting the next. It MAY start client transactions with every destination in parallel. It also MAY arbitrarily divide the set into groups, processing the groups serially and processing the destinations in each group in parallel. A common ordering mechanism is to use the qvalue parameter of destinations obtained from Contact header fields (see Section 24.10). Destinations are processed from highest qvalue to lowest. Destinations with equal qvalues may be processed in parallel. A stateful proxy must have a mechanism to maintain the destination set as responses are received and associate the responses to each forwarded request with the original request. For the purposes of this model, this mechanism is a "response context" created by the proxy layer before forwarding the first request. For each destination, the proxy forwards the request following these steps: 1. Make a copy of the received request 2. Update the Request-URI Various Authors [Page 90] Internet Draft SIP January 28, 2002 3. Add a Via header field value 4. Update the Max-Forwards field 5. Update the Route header field if present 6. Optionally add a Record-route header field value 7. Optionally add additional headers 8. send the new request 9. Set timer C Each of these steps is detailed below: 1. Copy request The proxy starts with a copy of the received request. The copy MUST initially contain all of the header fields from the received request. Only those fields detailed in the processing described below may be removed. The copy SHOULD maintain the ordering of the header fields as in the received request. The proxy MUST NOT reorder field values with a common field name (See Section 7.3.1). An actual implementation need not perform a copy; the primary requirement is that the processing of each next hop begin with the same request. 2. Request-URI The Request-URI in the copy's start line MUST be replaced with the URI for this destination. If the URI contains any parameters not allowed in a Request-URI, they MUST be removed. This is the essence of a proxy's role. This is the mechanism through which a proxy routes a request toward its destination. 3. Via The proxy MUST insert a Via header field into the copy before the existing Via header fields. The construction of this header follows the same guidelines of Section 8.1.1.7. This implies that the proxy will compute its own branch Various Authors [Page 91] Internet Draft SIP January 28, 2002 parameter, which will be globally unique for that branch, and contain the requisite magic cookie. Proxies choosing to detect loops have an additional constraint in the value they use for construction of the branch parameter. A proxy choosing to detect loops SHOULD create a branch parameter separable into two parts by the implementation. The first part MUST satisfy the constraints of Section 8.1.1.7 as described above. The second is used to perform loop detection and distinguish loops from spirals. Loop detection is performed by verifying that, when a request returns to a proxy, those fields having an impact on the processing of the request have not changed. The value placed in this part of the branch parameter SHOULD reflect all of those fields (including any Proxy-Require and Proxy-Authorization headers). This is to ensure that if the request is routed back to the proxy and one of those fields changes, it is treated as a spiral and not a loop (Section 16.3 item 2) A common way to create this value is to compute a cryptographic hash of the To, From, Call-ID header fields, the Request-URI of the request received (before translation) and the sequence number from the CSeq header field, in addition to any Proxy-Require and Proxy- Authorization fields that may be present. The algorithm used to compute the hash is implementation-dependent, but MD5 [21], expressed in hexadecimal, is a reasonable choice. (Base64 is not permissible for a token.) If a proxy wishes to detect loops, the "branch" parameter it supplies MUST depend on all information affecting processing of a request, including the incoming request-URI and any header values affecting the request's admission or routing. This is necessary to distinguish looped requests from requests whose routing parameters have changed before returning to this server. The request method MUST NOT be included in the calculation of the branch parameter. In particular, CANCEL and ACK requests (for non-2xx responses) MUST have the same branch value as the corresponding request they cancel or acknowledge. The branch parameter is used in correlating those requests at the server handling them (see Section 17.2.3 and 9.2). Various Authors [Page 92] Internet Draft SIP January 28, 2002 4. Max-Forwards If the copy does not contain a Max-Forwards header field, the proxy must add one with a field value of 70. Some existing UAs will not provide a Max-Forwards header field in a request. If the copy contains a Max-Forwards header field, the proxy must decrement its value by one (1). 5. Route If the copy contains a Route header field, the proxy's routing policy will determine whether that field should be modified. A proxy with a strict-routing policy MUST remove the first (topmost) Route header field value. (The strict- routing policy would have already placed that value into the Request-URI of this copy.) A proxy with a loose-routing policy MAY remove the topmost value. Restrictions on a loose-routing proxy's policy with respect to the topmost Route header are described in Section 8.1.3. 6. Record-Route If this proxy wishes remain on the path of future requests in a dialog created by this request, it MUST insert a Record-Route header value into the copy before any existing Record-Route header values, even if a Route field is already present. Requests establishing a dialog may contain preloaded Route header fields. If this request is already part of a dialog, the proxy SHOULD insert a Record-Route header field value if it wishes to remain on the path of future requests in the dialog. In normal endpoint operation as described in Section 12 these Record-Route header field values will not have any effect on the route sets used by the endpoints. The proxy will remain on the path if it choses to not insert a Record-Route header field value into requests that are already part of a dialog. However, it would be removed from the path when an endpoint that has Various Authors [Page 93] Internet Draft SIP January 28, 2002 failed reconstitutes the dialog. A proxy MAY insert a Record-Route header value into any request. If the request does not initiate a dialog, the endpoints will ignore the value. See Section 12 for details on how endpoints use the Record-Route header field values to construct Route header fields. Each proxy in the path of a request chooses whether to add a Record-Route header field value independently - the presence of a Record-Route header field in a request does not obligate this proxy to add a value. The URI placed in the Record-Route header value MUST be a SIP URI. This URI MAY be different for each destination the request is forwarded to. The URI SHOULD NOT contain the transport parameter unless the proxy has knowledge (such as in a private network) that the next downstream element that will be in the path of subsequent requests supports that transport. The URI this proxy provides will be used by some other element to make a routing decision. This proxy, in general, has no way to know what the capabilities of that element are, so it must restrict itself to the mandatory elements of a SIP implementation: SIP URIs and UDP transports. The URI placed in the Record-Route header value MUST resolve to this element when the server location procedures of [8] are applied to it. This ensures subsequent requests are routed back to this element. The URI placed in the Record-Route header value SHOULD be such that if a subsequent request is received with this URI in the Request-URI, the proxy's normal request processing will cause it to be forwarded to one of the previous elements, including the originating client, traversed by the original request. This improves robustness, ensuring that the Request-URI contains enough information to forward subsequent requests to a reasonable destination even in the absence of Route headers. The URI placed in the Record-Route header value MUST vary with the Request-URI in the received request. A request may legitimately pass through this proxy more than once on the way to its final destination (this is called a spiraling Various Authors [Page 94] Internet Draft SIP January 28, 2002 request). The Request-URI will be different each time the request passes through. If this proxy places the same URI in the Record-Route header field each time, subsequent requests will be rejected as looped requests. It is insufficient to simply copy the Request-URI from each request into the Record-Route header. Some modification, such as adding an maddr parameter, is necessary. URIs satisfying the above paragraphs can be constructed in many ways. One way is to use a URI that is nearly the same as the Contact header in the initial request (if present, else the From field), but with the maddr and port set to resolve to the proxy, and with a transaction identifier added to the user part of the request-URI (in order to meet the requirement that the URI in the Record-Route be different for each distinct Request-URI). A call stateful proxy could use a URI of the form sip:proxy.example.com and use information from the stored call state to meet the requirements. The proxy MAY include Record-Route header parameters in the value it provides. These will be returned in some responses to the request (200 (OK) responses to INVITE for example) and may be useful for pushing state into the message. The Record-Route process is designed to work for any SIP request that initiates a dialog. The only such request in this specification is INVITE. Extensions to the protocol MAY define others, and the mechanisms described here will apply. If a proxy needs to be in the path of any type of dialog (such as one straddling a firewall), it SHOULD add a Record-Route header value to every request with a method it does not understand since that method may have dialog semantics. The URI a proxy places into a Record-Route value is only valid for the lifetime of any dialog created by transaction in which it occurs. A dialog-stateful proxy, for example, MAY refuse to accept future requests with that value in the Request-URI after the dialog has terminated. Non-dialog- stateful proxies, of course, have no concept of when the dialog has terminated, but they MAY encode enough information in the value to compare it against the dialog identifier of future requests and MAY reject requests not matching that information. Endpoints MUST NOT use a URI obtained from a Record-Route header value outside the Various Authors [Page 95] Internet Draft SIP January 28, 2002 dialog in which it was provided. See Section 12 for more information on an endpoint's use of Record-Route header values. Generally, the choice about whether to record-route or not is a tradeoff of features vs. performance. Faster request processing and higher scalability is achieved when proxies do not record route. However, provision of certain services may require a proxy to observe all messages in a dialog. It is RECOMMENDED that proxies do not automatically record route. They should do so only if specifically required. 7. Adding Additional Headers The proxy MAY add any other appropriate headers to the copy at this point. 8. Forward Request A stateful proxy creates a new client transaction for this request as described in Section 17.1. If the next-hop location used in building this request contains the optional addressing parameters, the transaction is instructed to send the request based on those parameters. Otherwise, the proxy uses the procedures of Section [8] to compute an ordered set of addresses from the Request-URI, and as described there, attempts to contact the first one by instructing the client transaction to send the request there. If the client transaction reports failure to send the request or a timeout from its state machine, the stateful proxy continues to the next address that ordered set. Each attempt is a new client transaction, and therefore represents a new branch, so that the processing described above for each branch would need to be repeated. This results in a requirement to use a different branch ID parameter for each attempt. If the ordered set is exhausted, the request cannot be forwarded to this element in the destination set. The proxy does not need to place anything in the response context, but otherwise acts as if this element of the destination set returned a 408 (Request Timeout) final response. 9. Set timer C In order to handle the case where an INVITE request never generates a final response, a transaction timeout value is used. This is accomplished through a timer, called timer C, which MUST set for each client transaction when an INVITE Various Authors [Page 96] Internet Draft SIP January 28, 2002 request is proxied. The timer MUST be larger than 3 minutes. Section 16.6 bullet 2 discusses how this timer is updated with provisional responses, and Section 16.7 discusses processing when it fires. 16.6 Response Processing When a response is received by an element, it first tries to locate a client transaction (Section 17.1.3) matching the response. If none is found, the element MUST process the response (even if it is an informational response) as a stateless proxy (described below). If a match is found, the response is handed to the client transaction. Forwarding responses for which a client transaction (or more generally any knowledge of having sent an associated request) is not found improves robustness. In particular, it ensures that "late" 2xx class responses to INVITE requests are forwarded properly. As client transactions pass responses to the proxy layer, the following processing MUST take place: 1. Find the appropriate response context 2. Update timer C for provisional responses 3. Remove the topmost Via 4. Add the response to the response context 5. Check to see if this response should be forwarded The following processing MUST be performed on each response that is forwarded. It is likely that more than one response to each request will be forwarded: at least each provisional and one final response. 1. Aggregate authorization header fields if necessary; 2. forward the response; 3. generate any necessary CANCEL requests. If no final response has been forwarded after every client transaction associated with the response context has been terminated, the proxy must choose and forward the "best" response from those it has seen so far. Various Authors [Page 97] Internet Draft SIP January 28, 2002 Each of the above steps are detailed below: 1. Find Context The proxy locates the "response context" it created before forwarding the original request using the key described in Section 16.5. The remaining processing steps take place in this context. 2. Update timer C for provisional responses For an INVITE transaction, if the response is a provisional response with status codes 101 to 199 inclusive (i.e., anything but 100), the proxy MUST reset timer C for that client transaction. The timer MAY be reset to a different value, but this value MUST be greater than 3 minutes. 3. Via The proxy removes the topmost Via field value from the response. If no Via field values remain in the response, the response was meant for this element and MUST NOT be forwarded. The remainder of the processing described in this section is not performed on this message, the UAC processing rules described in Section 8.1.4 are followed instead (transport layer processing has already occurred). This will happen, for instance, when the element generates CANCEL requests as described in Section 10. 4. Add response to context ; Final responses received are stored in the response context until a final response is generated on the server transaction associated with this context. The response may be a candidate for the best final response to be returned on that server transaction. Information from this response may be needed in forming the best response even if this response is not chosen. If the proxy chooses to recurse on any contacts in a 3xx class response by adding them to the destination set, it MUST remove them from the response before adding the response to the response context. If the proxy recurses on all of the contacts in a 3xx class response, the proxy SHOULD NOT add the resulting contactless response to the Various Authors [Page 98] Internet Draft SIP January 28, 2002 response context. Removing the contact before adding the response to the response contact prevents the next element upstream from retrying a location this proxy has already attempted. 3xx class responses may contain a mixture of SIP and non- SIP URIs. A proxy may choose to recurse on the SIP URIs and place the remainder into the response context to be returned potentially in the final response. If a proxy receives a 416 (Unsupported URI Scheme) response to a request whose Request-URI scheme was not SIP, but the scheme in the original received request was SIP (that is, the proxy changed the scheme from SIP to something else when it proxied a request), the proxy SHOULD add a new URI to the destination set. This URI SHOULD be a SIP URI version of the non-SIP URI that was just tried. In the case of the tel URL, this is accomplished by placing the telephone-subscriber part of the tel URL into the user part of the SIP URI, and setting the hostpart to the domain where the prior request was sent. As with a 3xx response, if a proxy "recurses" on the 416 by trying a SIP URI instead, the 416 response SHOULD NOT be added to the response context. 5. Check response for forwarding Until a final response has been sent on the server transaction, the following responses MUST be forwarded immediately: - Any provisional response other than 100 (Trying) - Any 2xx response If a 6xx response is received, it is not immediately forwarded, but the stateful proxy SHOULD cancel all pending transactions as described in Section 10. This is a change from RFC 2543, which mandated that the proxy was to forward the 6xx response immediately. For an INVITE transaction, this approach had the problem that a 2xx response could arrive on another Various Authors [Page 99] Internet Draft SIP January 28, 2002 branch, in which case the proxy would have to forward the 2xx. The result was that the UAC could receive a 6xx response followed by a 2xx response, which should never be allowed to happen. Under the new rules, upon receiving a 6xx, a proxy will issue a CANCEL request, which will generally result in 487 responses from all outstanding client transactions, and then at that point the 6xx is forwarded upstream. After a final response has been sent on the server transaction, the following responses MUST be forwarded immediately: - Any 2xx class response to an INVITE request A stateful proxy MUST NOT immediately forward any other responses. In particular, a stateful proxy MUST NOT forward any 100 (Trying) response. Those responses that are candidates for forwarding later as the "best" response have been gathered as described in step "Add Response to Context". Any response chosen for immediate forwarding MUST be processed as described in steps "Aggregate authorization headers" through "Record-Route". This step, combined with the next, ensures that a stateful proxy will forward exactly one final response to a non- INVITE request, and either exactly one non-2xx class response or one or more 2xx-class responses to an INVITE request. 6. Choosing the best response A stateful proxy MUST send a final response to a response context's server transaction if no final responses have been immediately forwarded by the above rules and all client transactions in this response context have been terminated. The stateful proxy MUST choose the "best" final response among those received and stored in the response context. If there are no final responses in the context, the proxy MUST send a 408 (Request Timeout) response to the server transaction. Otherwise, the proxy MUST forward one of the responses from Various Authors [Page 100] Internet Draft SIP January 28, 2002 the lowest response class stored in the response context. The proxy MAY select any response within that lowest class. The proxy SHOULD give preference to responses that provide information affecting resubmission of this request, such as 401, 407, 415, 420, and 484. A proxy which receives a 503 (Service Unavailable) response SHOULD NOT forward it upstream unless it can determine that any subsequent requests it might proxy will also generate a 503. In other words, forwarding a 503 means that the proxy knows it cannot service any requests, not just the one for the Request-URI in the request which generated the 503. The forwarded response MUST be processed as described in steps "Aggregate authorization headers" through "Record- Route". For example, if a proxy forwarded a request to 4 locations, and received 503, 407, 501, and 404 responses, it may choose to forward the 407 (Proxy Authentication Required) response. 1xx and 2xx class responses may be involved in the establishment dialogs. When a request does not contain a To tag, the To tag in the response is used by the UAC to distinguish multiple responses to a dialog creating request. A proxy MUST NOT insert a tag into the To header of a 1xx or 2xx class response if the request did not contain one. A proxy MUST NOT modify the tag in the To header of a 1xx or 2xx class response. Since a proxy may not insert a tag into the To header of a 1xx class response to a request that did not contain one, it cannot issue non-100 provisional responses on its own. However, it can branch the request to a UAS sharing the same element as the proxy. This UAS can return its own provisional responses, entering into an early dialog with the initator of the request. The UAS does not have to be a discreet process from the proxy. It could be a virtual UAS implemented in the same code space as the proxy. 3-6xx class responses are delivered hop-hop. When issuing a 3-6xx class response, the element is effectivly acting as a UAS, issuing its own response, usually based on the responses received from downstream elements. An element SHOULD preserve the To tag when simply forwarding a 3-6xx class response to a request that did not contain a To tag. Various Authors [Page 101] Internet Draft SIP January 28, 2002 A proxy MUST NOT modify the To tag in any forwarded response to a request that contains a To tag. While it makes no difference to the upstream elements if the proxy replaced the To tag in a forwarded 3-6xx class response, preserving the original tag may assist with debugging. When the proxy is aggregating information from several responses, choosing a To tag from among them is arbitrary, and generating a new To tag may make debugging easier. This happens, for instance, when combining 401 (Unauthorized) and 407 (Proxy Authentication Required) challenges, or combining Contact values from unencrypted and unauthenticated 3xx class responses. 7. Aggregate authorization headers If the selected response is a 401 (Unauthorized) or 407 (Proxy Authentication Required), the proxy MUST collect any WWW-Authenticate and Proxy-Authenticate header fields from all other 401 (Unauthorized) and 407 (Proxy Authentication Required) responses received so far in this response context and add them to this response before forwarding. Each WWW-Authenticate and Proxy-Authenticate header field added to the response MUST preserve that header field value. The resulting 401 (Unauthorized) or 407 (Proxy Authenication Required) response may have several WWW- Authenticate AND Proxy-Authenticate headers. This is necessary because any or all of the destinations the request was forwarded to may have requested credentials. The client must receive all of those challenges and supply credentials for each of them when it retries the request. Motivation for this behavior is provided in Section 22. 8. Record-Route If the selected response contains a Record-Route header field value originally provided by this proxy, the proxy MAY chose to rewrite the value before forwarding the response. This allows the proxy to provide different URIs for itself to the next upstream and downstream elements. A proxy may choose to use this mechanism for any reason. For instance, it is useful for multi-homed hosts. Various Authors [Page 102] Internet Draft SIP January 28, 2002 The new URI provided by the proxy MUST satisfy the same constraints on URIs placed in Record-Route header fields in requests (see Step 6 of Section 16.5) with the following modifications: The URI SHOULD NOT contain the transport parameter unless the proxy has knowledge that the next upstream (as opposed to downstream) element that will be in the path of subsequent requests supports that transport. The URI placed in the Record-Route header value SHOULD be such that if a subsequent request is received with this URI in the Request-URI, the proxy's normal request processing will cause it to be forwarded to the same next-hop element (as opposed to some previous element) as the originally forwarded request. When a proxy does decide to modify the Record-Route header in the response, one of the operations it must perform is to locate the Record-Route that it had inserted. If the request spiraled, and the proxy inserted a Record-Route in each iteration of the spiral, locating the correct header in the response (which must be the proper iteration in the reverse direction) is tricky. The rules above dictate that a proxy insert a different URI into the Record-Route for each distinct Request-URI received. The two issues can be solved jointly. A RECOMMENDED mechanism is for the proxy to append a piece of data to the user portion of the URI. This piece of data is a hash of the transaction key (those peices of data used to match a request against existing transactions as discussed in section 17.2.3) for the incoming request, concatenated with a unique identifier for the proxy instance. Since the transaction key either contains Request-URI or depends on it (when the key is encoded in the branch parameter of the topmost Via header), this key will be unique for each distinct Request-URI. When the response arrives, the proxy modifies the first Record-Route whose identifier matches the proxy instance. The modification results in a URI without this piece of data appended to the user portion of the URI. Upon the next iteration, the same algorithm (find the topmost Record- Route header with the parameter) will correctly extract the next Record-Route header inserted by that proxy. 9. Forward response After performing the processing described in steps "Aggregate authorization headers" through "Record-Route", Various Authors [Page 103] Internet Draft SIP January 28, 2002 the proxy may perform any feature specific manipulations on the selected response. Unless otherwise specified, the proxy MUST NOT remove the message body or any header values other than the Via header value discussed in Section 3. In particular, the proxy MUST NOT remove any "received" parameter it may have added to the next Via header value while processing the request associated with this response. The proxy MUST pass the response to the server transaction associated with the response context. This will result in the response being sent to the location now indicated in the topmost Via field value. If the server transaction is no longer available to handle the transmission, the element MUST forward the response statelessly by sending it to the server transport. The server transaction may indicate failure to send the response or signal a timeout in its state machine. These errors should be logged for diagnostic purposes as appropriate, but the protocol requires no remedial action from the proxy. The proxy MUST maintain the response context until all of its associated transactions have been terminated, even after forwarding a final response. 10. Generate CANCELs OPEN ISSUE #7: If CANCEL is restricted to INVITE only, this behavior must restrict itself to INVITE requests. If the forwarded response was a final response, the proxy MUST generate a CANCEL request for all pending client transactions associated with this response context. A proxy SHOULD also generate a CANCEL request for all pending client transactions associated with this response context when it receives a 6xx response. A pending client transaction is one that has received a provisional response, but no final response and has not had an associated CANCEL generated for it. Generating CANCEL requests is described in Section 9.1. The requirement to CANCEL pending client transactions upon forwarding a final response does not guarantee that an endpoint will not receive multiple 200 (OK) responses to an INVITE. 200 (OK) responses on more than one branch may be generated before the CANCEL requests can be sent and processed. Further, it is reasonable to expect that a future extension may override this requirement to issue Various Authors [Page 104] Internet Draft SIP January 28, 2002 CANCEL requests. 16.7 Processing Timer C If timer C should fire, the proxy MUST either reset the timer with any value it chooses, or generate a CANCEL for that particular request. 16.8 Handling Transport Errors If the transport layer notifies a proxy of an error when it tries to forward a request (see Section 19.4), the proxy MUST behave as if the forwarded request received a 400 (Bad Request) response. If the proxy is notified of an error when forwarding a response, it drops the response. The proxy SHOULD NOT cancel any outstanding client transactions associated with this response context due to this notification. If a proxy cancels its outstanding client transactions, a single malicious or misbehaving client can cause all transactions to fail through its Via header field. 16.9 CANCEL Processing A stateful proxy may generate a CANCEL to any other request it has generated at any time (subject to receiving a provisional response to that request as described in section 9.1). A proxy MUST cancel any pending client transactions associated with a response context when it receives a matching CANCEL request. A stateful proxy MAY generate CANCEL requests for pending INVITE client transactions based on the period specified in the INVITEs Expires header field elapsing. However, this is generally unnecessary since the endpoints involved will take care of signaling the end of the transaction. While a CANCEL request is handled in a stateful proxy by its own server transaction, a new response context is not created for it. Instead, the proxy layer searches its existing response contexts for the server transaction handling the request associated with this CANCEL. If a matching response context is found, the element MUST immediately return a 200 (OK) response to the CANCEL request. In this case, the element is acting as a user agent server as defined in Section 8.2. Furthermore, the element MUST generate CANCEL requests for all pending client transactions in the context as described in Section 10. Various Authors [Page 105] Internet Draft SIP January 28, 2002 If a response context is not found, the element does not have any knowledge of the request to apply the CANCEL to. It MUST forward the CANCEL request (it may have statelessly forwarded the associated request previously). 16.10 Stateless Proxy When acting statelessly, a proxy is a simple message forwarder. Much of the processing performed when acting statelessly is the same as when behaving statefully. The differences are detailed here. A stateless proxy does not have any notion of a transaction, or of the response context used to describe stateful proxy behavior. Instead, the stateless proxy takes messages, both requests and responses, directly from the transport layer (See section 19). As a result, stateless proxies do not retransmit messages on their own. They do, however, forward all retransmission they receive (they do not have the ability to distinguish a retransmission from the original message). Furthermore, when handling a request statelessly, an element MUST NOT generate its own 100 (Trying) or any other provisional response. A stateless proxy must validate a request as described in Section 16.3 A stateless proxy must make a routing decision as described in Section 16.4 with the following exception: o A stateless proxy MUST choose one and only one destination from the destination set. This choice MUST only rely on fields in the message and time-invariant properties of the server. In particular, a retransmitted request MUST be forwarded to the same destination each time it is processed. Furthermore, CANCEL and non-Routed ACK requests MUST generate the same choice as their associated INVITE. A stateless proxy must process the request before forwarding as described in Section 16.5 with the following exceptions: o The requirement for unique branch IDs across time applies to stateless proxies as well. However, a stateless proxy cannot simply use a random number generator to compute the first component of the branch ID, as described in Section 16.5 bullet 3. This is because retransmissions of a request need to have the same value, and a stateless proxy cannot tell a retransmission from the original request. Therefore, the component of the branch parameter that makes it unique MUST be the same each time a retransmitted request is forwarded. Thus Various Authors [Page 106] Internet Draft SIP January 28, 2002 for a stateless proxy, the branch parameter MUST be computed as a combinatoric function of message parameters which are invariant on retransmission. o The stateless proxy MAY use any technique it likes to guarantee uniqueness of its branch IDs across transactions. However, the following procedure is RECOMMENDED. The proxy examines the branch ID of the received request. If it begins with the magic cookie, the first component of the branch ID of the outgoing request is computed as a hash of the received branch ID. Otherwise, the first component of the branch ID is computed as a hash of the topmost Via, the To header, the From header , the Call-ID header, the CSeq number (but not method), and the Request-URI from the received request. One of these fields will always vary across two different transactions. o The request is sent directly to the transport layer instead of through a client transaction. If the next-hop destination parameters don't provide an explicit destination, the element applies the procedures of [8] to the Request-URI to determine where to send the request. Since a stateless proxy must forward retransmitted requests to the same destination and add identical branch parameters to each of them, it can only use information from the message itself and time-invariant configuration data for those calculations. If the configuration state is not time-invariant (for example, if a routing table is updated) any requests that could be affected by the change may not be forwarded statelessly during an interval equal to the transaction timeout window before or after the change. The method of processing the affected requests in that interval is an implementation decision. A common solution is to forward them transaction statefully. Stateless proxies MUST NOT perform special processing for CANCEL requests. They are processed by the above rules as any other requests. In particular, a stateless proxy applies the same Route header processing to CANCEL requests that it applies to any other request. Response processing as described in Section 16.6 does not apply to a proxy behaving statelessly. When a response arrives at a stateless proxy, the proxy inspects the sent-by value in the first (topmost) Via header value. If that address matches the proxy (it equals a value this proxy has inserted into previous requests) the proxy MUST remove that value from the response and forward the result to the Various Authors [Page 107] Internet Draft SIP January 28, 2002 location indicated in the next Via header value. Unless specified otherwise, the proxy MUST NOT remove any other header values or the message body. If the address does not match the proxy, the message MUST be silently discarded. 16.11 Record-Route Example This example demonstrates one way Record-Route header values can be constructed to satisfy the requirements described in section 16.5 item 6 and section 16.6 item 8. Consider a proxy at server12.atlanta.com listening on port 5061 which receives the following request (many headers are omitted for brevity): INVITE sip:user@example.com SIP/2.0 Via: SIP/2.0/UDP callerspc.univ.edu Contact: sip:caller@callerspc.univ.edu The proxy forwards this request to a UAS at sip:j_user@div11.example.com, and record-routes: INVITE sip:j_user@div11.example.com SIP/2.0 Via: SIP/2.0/UDP server12.atlanta.com:5061 Via: SIP/2.0/UDP callerspc.univ.edu Record-Route: Contact: sip:caller@callerspc.univ.edu The 200 (OK) response received by the proxy will look like, in part: SIP/2.0 200 OK Via: SIP/2.0/UDP server12.atlanta.com:5061 Via: SIP/2.0/UDP callerspc.univ.edu Record-Route: Contact: sip:j_user@host32.div11.example.com The proxy modifies its Record-Route header in the response, resulting Various Authors [Page 108] Internet Draft SIP January 28, 2002 in the new response forwarded upstream: SIP/2.0 200 OK Via: SIP/2.0/UDP callerspc.univ.edu Record-Route: Contact: sip:j_user@host32.div11.example.com The route set computed by the UAS is: sip:caller.8jjs@callerspc.univ.edu:5061;maddr=server12.atlanta.com sip:caller@callerspc.univ.edu and the route set computed by the UAC is: sip:j_user@example.com:5061;maddr=server12.atlanta.com sip:j_user@host32.div11.example.com 17 Transactions SIP is a transactional protocol: interactions between components take place in a series of independent message exchanges. Specifically, a SIP transaction consists of a single request, and any responses to that request (which include zero or more provisional responses and one or more final responses). In the case of a transaction where the request was an INVITE (known as an INVITE transaction), the transaction also includes the ACK only if the final response was not a 2xx response. If the response was a 2xx, the ACK is not considered part of the transaction. The reason for this separation is rooted in the importance of delivering all 200 (OK) responses to an INVITE to the UAC. To deliver them all to the UAC, the UAS alone takes responsibility for retransmitting them, and the UAC alone takes responsibility for acknowledging them with ACK. Since this ACK is retransmitted only by the UAC, it is effectively considered its own transaction. Transactions have a client side and a server side. The client side is known as a client transaction, and the server side, as a server Various Authors [Page 109] Internet Draft SIP January 28, 2002 transaction. The client transaction sends the request, and the server transaction sends the response. The client and server transactions are logical functions that are embedded in any number of elements. Specifically, they exist within user agents and stateful proxy servers. Consider the example of Section 4. In this example, the UAC executes the client transaction, and its outbound proxy executes the server transaction. The outbound proxy also executes a client transaction, which sends the request to a server transaction in the inbound proxy. That proxy also executes a client transaction, which in turn, sends the request to a server transaction in the UAS. This is shown pictorially in Figure 4. +---------+ +---------+ +---------+ +---------+ | +-+|Request |+-+ +-+|Request |+-+ +-+|Request |+-+ | | |C||------->||S| |C||------->||S| |C||------->||S| | | |l|| ||e| |l|| ||e| |l|| ||e| | | |i|| ||r| |i|| ||r| |i|| ||r| | | |e|| ||v| |e|| ||v| |e|| ||v| | | |n|| ||e| |n|| ||e| |n|| ||e| | | |t|| ||r| |t|| ||r| |t|| ||r| | | | || || | | || || | | || || | | | |T|| ||T| |T|| ||T| |T|| ||T| | | |r|| ||r| |r|| ||r| |r|| ||r| | | |a|| ||a| |a|| ||a| |a|| ||a| | | |n|| ||n| |n|| ||n| |n|| ||n| | | |s||Response||s| |s||Response||s| |s||Response||s| | | +-+|<-------|+-+ +-+|<-------|+-+ +-+|<-------|+-+ | +---------+ +---------+ +---------+ +---------+ UAC Outbound Inbound UAS Proxy Proxy Figure 4: Transaction relationships A stateless proxy does not contain a client or server transaction. The transaction exists between the UA or stateful proxy on one side of the stateless proxy, and the UA or stateful proxy on the other Various Authors [Page 110] Internet Draft SIP January 28, 2002 side. As far as SIP transactions are concerned, stateless proxies are effectively transparent. The purpose of the client transaction is to receive a request from the element the client is embedded in (call this element the "Transaction User" or TU; it can be a UA or a stateful proxy), and reliably deliver the request to that server transaction. The client transaction is also responsible for receiving responses, and delivering them to the TU, filtering out any retransmissions or disallowed responses (such as a response to ACK). In the case of an INVITE transaction, that includes generation of the ACK request for any final response excepting a 2xx response. Similarly, the purpose of the server transaction is to receive requests from the transport layer, and deliver them to the TU. The server transaction filters any request retransmissions from the network. The server transaction accepts responses from the TU, and delivers them to the transport layer for transmission over the network. In the case of an INVITE transaction, it absorbs the ACK request for any final response excepting a 2xx response. The 2xx response, and the ACK for it, have special treatment. This response is retransmitted only by a UAS, and its ACK generated only by the UAC. This end-to-end treatment is needed so that a caller knows the entire set of users that have accepted the call. Because of this special handling, retransmissions of the 2xx response are handled by the UA core, not the transaction layer. Similarly, generation of the ACK for the 2xx is handled by the UA core. Each proxy along the path merely forwards each 2xx response to INVITE, and its corresponding ACK. A reliable provisional response, and the PRACK for it, also have special treatment. Reliable provisional responses are also only retransmitted by the UAS core, and the PRACK generated by the UAC core. Unlike ACK, however, PRACK is a normal non-INVITE transaction, which means that it will generate its own final response. The reason for this seemingly inexplicable difference between PRACK and ACK is that reliability of provisional responses was added on later as an extra feature, and therefore needed to be done within the confines of SIP extensibility. SIP extensibility only allowed the additions of new methods which behaved like any other non-INVITE method. 17.1 Client Transaction The client transaction provides its functionality through the maintenance of a state machine. The TU communicates with the client transaction through a simple interface. When the TU wishes to initiate a new transaction, it creates a client transaction, and passes it the SIP request to send, Various Authors [Page 111] Internet Draft SIP January 28, 2002 and an IP address, port, and transport to send it to. The client transaction begins execution of its state machine. Valid responses are passed up to the TU from the client transaction. There are two types of client transaction state machines, depending on the method of the request passed by the TU. One handles client transactions for INVITE request. This type of machine is referred to as an INVITE client transaction. Another type handles client transactions for all requests except INVITE and ACK. This is referred to as a non-INVITE client transaction. There is no client transaction for ACK. If the TU wishes to send an ACK, it passes one directly to the transport layer for transmission. The INVITE transaction is different from those of other methods because of its extended duration. Normally, human input is required in order to respond to an INVITE. The long delays expected for sending a response argue for a three way handshake. Requests of other methods, on the other hand, are expected to completely rapidly. In fact, because of its reliance on just a two way handshake, TUs SHOULD respond immediately to non-INVITE requests. Protocol extensions which require longer durations for generation of a response (such as a new method that does require human interaction) SHOULD instead use two transactions - one to send the request, and another in the reverse direction to convey the result of the request. 17.1.1 INVITE Client Transaction 17.1.1.1 Overview of INVITE Transaction The INVITE transaction consists of a three-way handshake. The client transaction sends an INVITE, the server transaction sends responses, and the client transaction sends an ACK. For unreliable transports (such as UDP), the client transaction will retransmit requests at an interval that starts at T1 seconds and doubles after every retransmission. T1 is an estimate of the RTT, and it defaults to 500 ms. Nearly all of the transaction timers described here scale with T1, and changing T1 is how their values are adjusted. The request is not retransmitted over reliable transports. After receiving a 1xx response, any retransmissions cease altogether, and the client waits for further responses. The server transaction can send additional 1xx responses, which are not transmitted reliably by the server transaction. If the provisional response needs to be sent reliably, this is handled by the TU. Eventually, the server transaction decides to send a final response. For unreliable transports, that response is retransmitted periodically, and for reliable transports, its sent once. For each final response that is received at the client transaction, the client transaction sends an ACK, the purpose of which is to quench retransmissions of the response. Various Authors [Page 112] Internet Draft SIP January 28, 2002 17.1.1.2 Formal Description The state machine for the INVITE client transaction is shown in Figure 5. The initial state, "calling", MUST be entered when the TU initiates a new client transaction with an INVITE request. The client transaction MUST pass the request to the transport layer for transmission (see Section 19). If an unreliable transport is being used, the client transaction SHOULD start timer A with a value of T1, and SHOULD NOT start timer A when a reliable transport is being used (Timer A controls request retransmissions). For any transport, the client transaction MUST start timer B with a value of 64*T1 seconds (Timer B controls transaction timeouts). When timer A fires, the client transaction SHOULD retransmit the request by passing it to the transport layer, and SHOULD reset the timer with a value of 2*T1. The formal definition of retransmit within the context of the transaction layer, is to take the message previously sent to the transport layer, and pass it to the transport layer once more. When timer A fires 2*T1 seconds later, the request SHOULD be retransmitted again (assuming the client transaction is still in this state). This process SHOULD continue, so that the request is retransmitted with intervals that double after each transmission. These retransmissions SHOULD only be done while the client transaction is in the "calling" state. The default value for T1 is 500 ms. T1 is an estimate of the RTT between the client and server transactions. The optional RTT estimation procedure of Section 17.3 MAY be followed, in which case the resulting estimate MAY be used instead of 500 ms. If no RTT estimation is used, other values MAY be used in private networks where it is known that RTT has a different value. On the public Internet, T1 MAY be chosen larger, but SHOULD NOT be smaller. If the client transaction is still in the "calling"state when timer B fires, the client transaction SHOULD inform the TU that a timeout has occurred. The client transaction MUST NOT generate an ACK. The value of 64*T1 is equal to the amount of time required to send seven requests in the case of an unreliable transport. If the client transaction receives a provisional response while in the "calling" state, it transitions to the "proceeding" state. In the "proceeding" state, the client transaction SHOULD NOT retransmit the request any longer. Furthermore, the provisional response MUST be passed to the TU. Any further provisional responses MUST be passed up to the TU while in the "proceeding" state. Passing of all provisional Various Authors [Page 113] Internet Draft SIP January 28, 2002 responses is necessary since the TU will handle reliability of these messages, and therefore even retransmissions of a provisional response must be passed upwards. When in either the "calling" or "proceeding" states, reception of a response with status code from 300-699 MUST cause the client transaction to transition to "completed". The client transaction MUST pass the received response up to the TU, and the client transaction MUST generate an ACK request, even if the transport is reliable (guidelines for constructing the ACK from the response are given in Section 17.1.1.3) and then pass the ACK to the transport layer for transmission. The ACK MUST be sent to the same address, port and transport that the original request was sent to. The client transaction SHOULD start timer D when it enters the "completed" state, with a value of at least 32 seconds for unreliable transports, and a value of zero seconds for reliable transports. Timer D is a reflection of the amount of time that the server transaction can remain in the "completed" state when unreliable transports are used. This is equal to Timer H in the INVITE server transaction, whose default is 64*T1. However, the client transaction does not know the value of T1 in use by the server transaction, so an absolute minimum of 32s is used instead of basing Timer D on T1. Any retransmissions of the final response that are received while in the "completed" state SHOULD cause the ACK to be re-passed to the transport layer for retransmission, but the newly received response MUST NOT be passed up to the TU. A retransmission of the response is defined as any response which would match the same client transaction, based on the rules of Section 17.1.3. If timer D fires while the client transaction is in the "completed" state, the client transaction MUST move to the terminated state, and it MUST inform the TU of the timeout. When in either the "calling" or "proceeding" states, reception of a 2xx response MUST cause the client transaction to enter the terminated state, and the response MUST be passed up to the TU. The handling of this response depends on whether the TU is a proxy core or a UAC core. A UAC core will handle generation of the ACK for this response, while a proxy core will always forward the 200 (OK) upstream. The differing treatment of 200 (OK) between proxy and UAC is the reason that handling of it does not take place in the transaction layer. The client transaction MUST be destroyed the instant it enters the terminated state. This is actually necessary to guarantee correct operation. The reason is that 2xx responses to an INVITE are treated differently; each one is forwarded by proxies, and the ACK handling Various Authors [Page 114] Internet Draft SIP January 28, 2002 |INVITE from TU Timer A fires |INVITE sent Reset A, V Timer B fires INVITE sent +-----------+ or Transport Err. +---------| |---------------+inform TU | | Calling | | +-------->| |-------------->| +-----------+ 2xx | | | 2xx to TU | | |1xx | 300-699 +---------------+ |1xx to TU | ACK sent | | | resp. to TU | 1xx V | | 1xx to TU -----------+ | | +---------| | | | | |Proceeding |-------------->| | +-------->| | 2xx | | +-----------+ 2xx to TU | | 300-699 | | | ACK sent, | | | resp. to TU| | | | | NOTE: | 300-699 V | | ACK sent +-----------+Transport Err. | transitions | +---------| |Inform TU | labeled with | | | Completed |-------------->| the event | +-------->| | | over the action | +-----------+ | to take | ^ | | | | | Timer D fires | +--------------+ | - | | | V | +-----------+ | | | | | Terminated|<--------------+ | | +-----------+ Figure 5: INVITE client transaction Various Authors [Page 115] Internet Draft SIP January 28, 2002 in a UAC is different. Thus, each 2xx needs to be passed to a proxy core (so that it can be forwarded) and to a UAC core (so it can be acknowledged). No transaction layer processing takes place. Whenever a response is received by the transport, if the transport layer finds no matching client transaction (using the rules of Section 17.1.3), the response is passed directly to the core. Since the matching client transaction is destroyed by the first 2xx, subsequent 2xx will find no match and therefore be passed to the core. 17.1.1.3 Construction of the ACK Request The ACK request constructed by the client transaction MUST contain values for the Call-ID, From, and Request-URI which are equal to the values of those headers in the request passed to the transport by the client transaction (call this the "original request"). The To field in the ACK MUST equal the To field in the response being acknowledged, and will therefore usually differ from the To field in the original request by the addition of the tag parameter. The ACK MUST contain a single Via header, and this MUST be equal to the top Via header of the original request. The ACK request MUST contain the same Route headers as the request whose response it is acknowledging . The CSeq header in the ACK MUST contain the same value for the sequence number as was present in the original request, but the method parameter MUST be equal to "ACK". If the INVITE request whose response is being acknowledged had Route headers, those headers MUST appear in the ACK. This is to ensure that the ACK can be routed properly through any downstream stateless proxies. Although any request MAY contain a body, a body in an ACK is special since the request cannot be rejected if the body is not understood. Therefore, placement of bodies in ACK for non-2xx is NOT RECOMMENDED, but if done, the body types are restricted to any that appeared in the INVITE, assuming that that the response to the INVITE was not 415. If it was, the body in the ACK MAY be any type listed in the Accept header in the 415. These rules for construction of ACK only apply to the client transaction. A UAC core which generates an ACK for 2xx MUST instead follow the rules described in Section 13. For example, consider the following request: INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKkjshdyff To: Bob Various Authors [Page 116] Internet Draft SIP January 28, 2002 From: Alice ;tag=88sja8x Call-ID: 987asjd97y7atg CSeq: 986759 INVITE The ACK request for a non-2xx final response to this request would look like this: ACK sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKkjshdyff To: Bob ;tag=99sa0xk From: Alice ;tag=88sja8x Call-ID: 987asjd97y7atg CSeq: 986759 ACK 17.1.2 non-INVITE Client Transaction 17.1.2.1 Overview of the non-INVITE Transaction Non-INVITE transactions do not make use of ACK. They are a simple request-response interaction. For unreliable transports, requests are retransmitted at an interval which starts at T1, and doubles until it hits T2. If a provisional response is received, retransmissions continue for unreliable transports, but at an interval of T2. The server transaction retransmits the last response it sent (which can be a provisional or final response) only when a retransmission of the request is received. This is why request retransmissions need to continue even after a provisional response, they are what ensure reliable delivery of the final response. Unlike an INVITE transaction, a non-INVITE transaction has no special handling for the 2xx response. The result is that only a single 2xx response to a non-INVITE is ever delivered to a UAC. 17.1.2.2 Formal Description The state machine for the non-INVITE client transaction is shown in Figure 6. It is very similar to the state machine for INVITE. The "Trying" state is entered when the TU initiates a new client transaction with a request. When entering this state, the client transaction SHOULD set timer F to fire in 64*T1 seconds. The request MUST be passed to the transport layer for transmission. If an Various Authors [Page 117] Internet Draft SIP January 28, 2002 unreliable transport is in use, the client transaction MUST set timer E to fire in T1 seconds. If timer E fires while still in this state, the timer is reset, but this time with a value of MIN(2*T1, T2). When the timer fires again, it is reset to a MIN(4*T1, T2). This process continues, so that retransmissions occur with an exponentially increasing inverval that caps at T2. The default value of T2 is 4s, and it represents the amount of time a non-INVITE server transaction will take to respond to a request, if it does not respond immediately. For the default values of T1 and T2, this results in intervals of 500 ms, 1 s, 2 s, 4 s, 4 s, 4s, etc. If Timer F fires while the client transaction is still in the "Trying" state, the client transaction SHOULD inform the TU about the timeout, and then it SHOULD enter the "Terminated" state. If a provisional response is received while in the "Trying" state, the response MUST be passed to the TU, and then the client transaction SHOULD move to the "Proceeding" state. If a final response (status codes 200-699) is received while in the "Trying" state, the response MUST be passed to the TU, and the client transaction MUST transition to the "Completed" state. If Timer E fires while in the "Proceeding" state, the request MUST be passed to the transport layer for retransmission, and Timer E MUST be reset with a value of T2 seconds. If timer F fires while in the "Proceeding" state, the TU MUST be informed of a timeout, and the client transaction MUST transition to the terminated state. If a final response (status codes 200-699) is received while in the "Proceeding" state, the response MUST be passed to the TU, and the client transaction MUST transition to the "Completed" state. Once the client transaction enters the "Completed" state, it MUST set Timer K to fire in T4 seconds for unreliable transports, and zero seconds for reliable transports. The "Completed" state exists to buffer any additional response retransmissions that may be received (which is why the client transaction remains there only for unreliable transports). T4 represents the amount of time the network will take to clear messages between client and server transactions. The default value of T4 is 5s. A response is a retransmission when it matches the same transaction, using the rules specified in Section 17.1.3. If Timer K fires while in this state, the client transaction MUST transition to the "Terminated" state. Once the transaction is in the terminated state, it MUST be destroyed. As with client transactions, this is needed to ensure reliability of the 2xx responses to INVITE. 17.1.3 Matching Responses to Client Transactions Various Authors [Page 118] Internet Draft SIP January 28, 2002 |Request from app |send request Timer E V send request +-----------+ +---------| |-------------------+ | | Trying | Timer F | +-------->| | or Transport Err.| +-----------+ inform TU | 200-699 | | | resp. to TU | |1xx | +---------------+ |resp. to TU | | | | | Timer E V Timer F | | send req +-----------+ or Transport Err. | | +---------| | inform TU | | | |Proceeding |------------------>| | +-------->| |-----+ | | +-----------+ |1xx | | | ^ |resp to TU | | 200-699 | +--------+ | | resp. to TU | | | | | | V | | +-----------+ | | | | | | | Completed | | | | | | | +-----------+ | | ^ | | | | | Timer K | +--------------+ | - | | | V | NOTE: +-----------+ | | | | transitions | Terminated|<------------------+ labeled with | | the event +-----------+ over the action to take Figure 6: non-INVITE client transaction Various Authors [Page 119] Internet Draft SIP January 28, 2002 When the transport layer in the client receives a response, it has to figure out which client transaction will handle the response, so that the processing of Sections 17.1.1 and 17.1.2 can take place. The branch parameter in the top Via header is used for this purpose. A response matches a client transaction under two conditions. First, if the response has the same value of the branch parameter in the top Via header as the branch parameter in the top Via header of the request that created the transaction. Second, if the method parameter in the CSeq header matches the method of the request that created the transaction. The method is needed since a CANCEL request constitutes a different transaction, but shares the same value of the branch parameter. A response which matches a transaction matched by a previous response is considered a retransmission of that response. 17.1.4 Handling Transport Errors When the client transaction sends a request to the transport layer to be sent, the following procedures are followed if the transport layer indicates a failure. The client transaction SHOULD inform the TU that a transport failure has occurred, and the client transaction SHOULD transition directly to the terminated state. 17.2 Server Transaction The server transaction is responsible for the delivery of requests to the TU, and the reliable transmission of responses. It accomplishes this through a state machine. Server transactions are created by the core when a request is received, and transaction handling is desired for that request (this won't always be the case). As with the client transactions, the state machine depends on whether the received request is an INVITE request or not. 17.2.1 INVITE Server Transaction The state diagram for the INVITE server transaction is shown in Figure 7. When a server transaction is constructed with a request, it enters the "Proceeding" state. The server transaction MUST generate a 100 response (not any status code -- the specific value of 100) unless it knows that the TU will generate a provisional or final response Various Authors [Page 120] Internet Draft SIP January 28, 2002 withpin 200 ms, in which case it MAY generate a 100 (Trying) response. This provisional response is needed to rapidly quench request retransmissions in order to avoid network congestion. The 100 response is constructed according to the procedures in Section 8.2.6, except that insertion of tags in the To field of the response (when none was present in the request), is downgraded from MAY to SHOULD NOT. The request MUST be passed to the TU. The TU passes any number of provisional responses to the server transaction. So long as the server transaction is in the "Proceeding" state, each of these MUST be passed to the transport layer for transmission. They are not sent reliably by the transaction layer (they are not retransmitted by it), and do not cause a change in the state of the server transaction. When provisional responses need to be delivered reliably, it is handled by the TU, which will retransmit the provisional responses itself, and pass downwards each retransmission to the server transaction. If a request retransmission is received while in the "Proceeding" state, the most recent provisional response that was received from the TU MUST be passed to the transport layer for retransmission. A request is a retransmission if it matches the same server transaction based on the rules of Section 17.2.3. If, while in the "proceeding" state, the TU passes a 2xx Response to the server transaction, the server transaction MUST pass this response to the transport layer for transmission. It is not retransmitted by the server transaction; retransmissions of 2xx responses are handled by the TU. The server transaction MUST then transition to the "terminated" state. While in the "Proceeding" state, if the TU passes a response with status code from 300 to 699 to the server transaction, the response MUST be passed to the transport layer for transmission, and the state machine MUST enter the "Completed" state. For unreliable transports, timer G is set to fire in T1 seconds, and is not set to fire for reliable transports. This is a change from RFC 2543, where responses were always retransmitted, even over reliable transports. When the "Completed" state is entered, timer H MUST be set to fire in 64*T1 seconds, for all transports. Timer H determines when the server transaction gives up retransmitting the response. Its value is chosen to equal Timer B, the amount of time a client transaction will continue to retry sending a request. If timer G fires, the response is passed to the transport layer once more for retransmission, and timer G is set to fire in MIN(2*T1, T2) seconds. From then on, when Various Authors [Page 121] Internet Draft SIP January 28, 2002 |INVITE |pass INV to TU INVITE V send 100 if TU won't in 200ms send response+-----------+ +--------| |--------+101-199 from TU | | Proceeding| |send response +------->| |<-------+ | | Transport Err. | | Inform TU | |--------------->+ +-----------+ | 300-699 from TU | |2xx from TU | send response | |send response | | +------------------>+ | | INVITE V Timer G fires | send response+-----------+ send response | +--------| |--------+ | | | Completed | | | +------->| |<-------+ | +-----------+ | | | | ACK | | | - | +------------------>+ | Timer H fires | V or Transport Err.| +-----------+ Inform TU | | | | | Confirmed | | | | | +-----------+ | | | |Timer I fires | |- | | | V | +-----------+ | | | | | Terminated|<---------------+ | | +-----------+ Figure 7: INVITE server transaction Various Authors [Page 122] Internet Draft SIP January 28, 2002 timer G fires, the response is passed to the transport again for transmission, and timer G is reset with a value that doubles, unless that value exceeds T2, in which case it is reset with the value of T2. This is identical to the retransmit behavior for requests in the "Trying" state of the non- INVITE client transaction. Furthermore, while in the "completed" state, if a request retransmission is received, the server SHOULD pass the response to the transport for retransmission. If an ACK is received while the server transaction is in the "Completed" state, the server transaction MUST transition to the "confirmed" state. As Timer G is ignored in this state, any retransmissions of the response will cease. If timer H fires while in the "Completed" state, it implies that the ACK was never received. In this case, the server transaction MUST transition to the terminated state, and MUST indicate to the TU that a transaction failure has occurred. The purpose of the "confirmed" state is to absorb any additional ACK messages that arrive, triggered from retransmissions of the final response. When this state is entered, timer I is set to fire in T4 seconds for unreliable transports, and zero seconds for reliable transports. Once timer I fires, the server MUST transition to the "Terminated" state. Once the transaction is in the terminated state, it MUST be destroyed. As with client transactions, this is needed to ensure reliability of the 2xx responses to INVITE. 17.2.2 non-INVITE Server Transaction The state machine for the non-INVITE server transaction is shown in Figure 8. The state machine is initialized in the "Trying" state, and is passed a request other than INVITE or ACK when initialized. This request is passed up to the TU. Once in the "Trying" state, any further request retransmissions are discarded. A request is a retransmission if it matches the same server transaction, using the rules specified in Section 17.2.3. While in the "Trying" state, if the TU passes a provisional response to the server transaction, the server transaction MUST enter the "Proceeding" state. The response MUST be passed to the transport layer for transmission. Any further provisional responses that are received from the TU while in the "Proceeding" state MUST be passed Various Authors [Page 123] Internet Draft SIP January 28, 2002 to the transport layer for transmission. If a retransmission of the request is received while in the "Proceeding" state, the most recently sent provisional response MUST be passed to the transport layer for retransmission. If the TU passes a final response (status codes 200-699) to the server while in the "Proceeding" state, the transaction MUST enter the "Completed" state, and the response MUST be passed to the transport layer for transmission. When the server transaction enters the "Completed" state, it MUST set Timer J to fire in 64*T1 seconds for unreliable transports, and zero seconds for reliable transports. While in the "Completed" state, the server transaction MUST pass the final response to the transport layer for retransmission whenever a retransmission of the request is received. Any other final responses passed by the TU to the server transaction MUST be discarded while in the "Completed" state. The server transaction remains in this state until Timer J fires, at which point it MUST transition to the "Terminated" state. The server transaction MUST be destroyed the instant it enters the "Terminated" state. 17.2.3 Matching Requests to Server Transactions When a request is received from the network by the server, it has to be matched to an existing transaction. This is accomplished in the following manner. The branch parameter in the topmost Via header the request is examined. If it is present, and begins with the magic cookie "z9hG4bK", the request was generated by a client transaction compliant to this specification. Therefore, the branch parameter will be unique across all transactions sent by that client. The request matches a transaction if the branch parameter in the request is equal to the one in the top Via header of the request that created the transaction, the source address and port of the request are the same as the source address and port of the the request that created the transaction, and in the case of a CANCEL request, the method of the request that created the transaction was also CANCEL. This matching rule applies to both INVITE and non-INVITE transactions alike. Source address and port are used as part of the matching process because there could be duplication of branch parameters from different clients; uniqueness in time is mandated for construction of the parameter, but not uniqueness in space. If the branch parameter in the top Via header is not present, or does Various Authors [Page 124] Internet Draft SIP January 28, 2002 |Request received |pass to TU V +-----------+ | | | Trying |-------------+ | | | +-----------+ |200-699 from TU | |send response |1xx from TU | |send response | | | Request V 1xx from TU | send response+-----------+send response| +--------| |--------+ | | | Proceeding| | | +------->| |<-------+ | +<--------------| | | |Trnsprt Err +-----------+ | |Inform TU | | | | | | |200-699 from TU | | |send response | | Request V | | send response+-----------+ | | +--------| | | | | | Completed |-------------+ | +------->| | +<--------------| | |Trnsprt Err +-----------+ |Inform TU | | |Timer J fires | |- | | | V | +-----------+ | | | +-------------->| Terminated| | | +-----------+ Figure 8: non-INVITE server transaction Various Authors [Page 125] Internet Draft SIP January 28, 2002 not contain the magic cookie, the following procedures are used. These exist to handle backwards compatibility with RFC 2543 compliant implementations. The INVITE request matches a transaction if the Request-URI, To, From, Call-ID, CSeq, and top Via header match those of the INVITE request which created the transaction. In this case, the INVITE is a retransmission of the original one that created the transaction. The ACK request matches a transaction if the Request-URI, From, Call-ID, CSeq number (not the method), and top Via header match those of the INVITE request which created the transaction, and the To field of the ACK matches the To field of the response sent by the server transaction (which then includes the tag). Matching is done based on the matching rules defined for each of those headers. The usage of the tag in the To field helps disambiguate ACK for 2xx from ACK for other responses at a proxy which may have forwarded both responses (which can occur in unusual conditions). An ACK request that matches an INVITE transaction matched by a previous ACK is considered a retransmission of that previous ACK. For all other request methods, a request is matched to a transaction if the Request-URI, To, From, Call-ID and Cseq (including the method) and top Via header match those of the request which created the transaction. Matching is done based on the matching rules defined for each of those headers. When a non-INVITE request matches an existing transaction, it is a retransmission of the request which created that transaction. Because the matching rules include the Request-URI, the server cannot match a response to a transaction. When the TU passes a response to the server transaction, it must pass it to the specific server transaction for which the response is targeted. 17.2.4 Handling Transport Errors When the server transaction sends a response to the transport layer to be sent, the following procedures are followed if the transport layer indicates a failure. First, the procedures in [8] are followed, which attempt to deliver the response to a backup. If those should all fail, such that all elements generate ICMP errors, or no SRV records are present, the server transaction SHOULD inform the TU that a failure has occurred, and SHOULD transition to the terminated state. 17.3 RTT Estimation Most of the timeouts used in the transaction state machines derive Various Authors [Page 126] Internet Draft SIP January 28, 2002 from T1, which is an estimate of the RTT between the client and server transactions. This subsection defines optional procedures that a client can use to build up estimates of the RTT to a particular IP address. To perform this procedure, the client MUST maintain a table of variables for each destination IP address to which an RTT estimate is being made. If a client wishes to measure RTT for a particular IP address, it MUST include a Timestamp header into a request containing the time when the request is initially created and passed to a new client transaction, which transmits the request. If a 100 (Trying) response (not any 1xx, only the 100 (Trying) response) is received before the client transaction generates a retransmission, an RTT estimate is made. This is consistent with the RFC 2988 requirements on TCP for using Karn's algorithm in RTT estimation. The estimate, called R, is made by computing the difference between the current time and the value of Timestamp header in the 100 response. The value of R is applied to the estimation of RTO as described in Section 2 of RFC 2988 [22], with the following differences. First, the initial value of RTO is 500 ms for SIP, not 3 s as is used for TCP. Second, there is no minimum value for the RTO, as there is for TCP, if SIP is being run on a private network. When run on the public Internet, the minimum is 500 ms, as opposed to 1 s for TCP. This difference is because of the expected usage of SIP in private networks where rapid call setup times are service critical. Once RTO is computed, the timer T1 is set to the value of RTO, and all other timers scale proportionally as described above. This value of T1 would be used for scaling all of the client and server transaction timers described above, when a request or response, respectively, is sent to that IP address. If the IP address is that of a stateless proxy, the actual round trip time that is measured will be the average to all transaction stateful proxies or UAs that are reached through the stateless proxy. This estimate may therefore be too low or too high for a specific transactional element being communicated with through the stateless proxy. 18 Reliability of Provisional Responses Normally, provisional responses are not transmitted reliably. The TU generates a single provisional response, and passes it to the server transaction, which sends it once. RFC 2543 provided no means for reliable transmission of these messages. It was later observed that reliability was important in several Various Authors [Page 127] Internet Draft SIP January 28, 2002 cases, including interoperability scenarios with the PSTN. Therefore, an optional capability was added in this specification to support reliable transmission of provisional responses. The reliability mechanism works by mirroring the current reliability mechanisms for 2xx final responses to INVITE. Those requests are transmitted periodically by the TU, until a separate transaction, ACK, is received, that indicates reception of the 2xx by the UAC. The reliability for the 2xx to INVITE and ACK messages are end-to-end. In order to achieve reliability for provisional responses, we do nearly the same thing. Reliable provisional responses are retransmitted by the TU with an exponential backoff. Those retransmisions cease when a PRACK message is received. The PRACK request plays the same role as ACK, but for provisional responses. There is an important difference, however. PRACK is a normal SIP message, like BYE. As such, its own reliability is ensured hop-by-hop through each stateful proxy. Similarly, PRACK has its own response. If this were not the case, the PRACK message could not traverse existing proxy servers. Each provisional response is given a sequence number, carried in the RSeq header in the response. The PRACK messages contain an RAck header, which indicates the sequence number of the provisional response which is being acknowledged. The acknowledgements are not cumulative, and the specifications recommend a single outstanding provisional response at a time, for purposes of congestion control. 18.1 UAS Behavior A UAS MAY send any non-100 provisional response to INVITE reliably, so long as the initial INVITE request (the request whose provisional response is being sent reliably) contained a Supported header with the option tag 100rel specification does not allow reliable provisional responses for any method but INVITE, extensions that define new methods which can establish dialogs may make use of the mechanism. The UAS MUST send any non-100 provisional response reliably if the initial request contained a Require header with the option tag 100rel initial request with a 420 (Bad Extension) and include a Unsupported header containing the option tag 100rel A UAS MUST NOT attempt to send a 100 (Trying) response reliably. Only provisional responses numbered 101 to 199 may be sent reliably. If the request did not include either a Supported or Require header indicating this feature, the UAS MUST NOT send the provisional response reliably. Various Authors [Page 128] Internet Draft SIP January 28, 2002 100 (Trying) responses are hop-by-hop only. For this reason, the reliability mechanisms described here, which are end-to-end, cannot be used. An element which can act as a proxy can also send reliable provisional responses; in that case, it acts as a UAS for purposes of that transaction. However, it MUST NOT attempt to do so for any request that contains a tag in the To field. That is, a proxy cannot generate reliable provisional responses to requests sent within the context of a dialog. Of course, unlike a UAS, when the proxy element receives a PRACK that does not match any outstanding reliable provisional response, the PRACK MUST be proxied. The rest of this discussion assumes that the initial request contained a Supported or Require header listing 100rel , and that there is a provisional response to be sent reliably. The provisional response to be sent reliably is constructed by the UAS core according to the procedures of Section 8.2.6 and Section 12. Specifically, the provisional response MUST establish a dialog if one is not yet created. In addition, it MUST contain Require header containing the option tag 100rel , and MUST include an RSeq header. The value of the header for the first reliable provisional response in a transaction MUST be between 1 and 2**31 - 1. It is RECOMMENDED that it be chosen uniformly in this range. The RSeq numbering space is within a single transaction. This means that provisional responses for different requests MAY use the same values for the RSeq number. The reliable provisional response is passed to the transaction layer periodically with an interval that starts at T1 seconds and doubles for each retransmission (T1 is defined in Section 17). Once passed to the server transaction, it is added to an internal list of unacknowledged reliable provisional responses. This differs from retransmissions of 2xx responses, which cap at T2 seconds. This is because retransmissions of ACK are triggered on receipt of a 2xx, but retransmissions of PRACK take place independently of reception of 1xx. Retransmissions cease when a matching PRACK is received. PRACK is like any other request within a dialog, and the UAS core processes it according to the procedures of Sections 8.2 and 12.2.2. A matching PRACK is defined as one within the same dialog as the response, and whose method, CSeq-num, and response-num in the RAck header match, respectively, the method and sequence number from the CSeq and sequence number from the RSeq of the reliable provisional response. Various Authors [Page 129] Internet Draft SIP January 28, 2002 If a PRACK request is received that does not match any unacknowledged reliable provisional response, the UAS MUST respond to the PRACK with a 481 response. If the PRACK does match an unacknowledged reliable provisional response, it MUST be responded to with a 2xx response. The UAS can be certain at this point that the provisional response has been received in order. It SHOULD cease retransmissions of the reliable provisional response, and MUST remove it from the list of unacknowledged provisional responses. If a reliable provisional response is retransmitted for 64*T1 seconds without reception of a corresponding PRACK, the UAS SHOULD reject the original request with a 5xx response. If the PRACK contained a body, the body is treated in the same way a body in an ACK is treated. After the first reliable provisional response for a request has been acknowledged, the UAS MAY send additional reliable provisional responses. The UAS MUST NOT send a second reliable provisional response until the first is acknowledged. After the first, it is RECOMMENDED that the UAS not send an additional reliable provisional response until the previous is acknowledged. The first reliable provisional response receives special treatment because it conveys the intitial sequence number. If additional reliable provisional responses were sent before the first was acknowledged, the UAS could not be certain these were received in order. The value of the RSeq in each subsequent reliable provisional response for the same request MUST be greater by exactly one. RSeq numbers MUST NOT wrap around. Because the initial one is chosen to be less than 2**31 - 1, but the maximum is 2**32 - 1, there can be up to 2**31 reliable provisional responses per request, which is more than sufficient. Note that the UAS MAY send a final response to the initial request before having received PRACKs for all unacknowledged reliable provisional responses. In that case, it SHOULD NOT continue to retransmit the unacknowledged reliable provisional responses, but it MUST be prepared to process PRACK requests for those outstanding responses. A UAS MUST NOT send new reliable provisional responses (as opposed to retransmissions of unacknowledged ones) after sending a final response to a request. 18.2 UAC Behavior If a provisional response is received for the initial request, and that response contains a Require header containing the option tag 100rel , the response is to be sent reliably. If the response is a Various Authors [Page 130] Internet Draft SIP January 28, 2002 100 (Trying) (as opposed to 101 to 199), this option tag MUST be ignored, and the procedures below MUST NOT be used. Assuming the response is to be transmitted reliably, the UAC MUST create a new request with method PRACK. This request is sent within the dialog associated with the provisional response (indeed, the provisional response may have created the dialog). PRACK requests MAY contain bodies, which are interpreted according to their type and disposition. Note that the PRACK is like any other non-INVITE request within a dialog. In particular, a UAC SHOULD NOT retransmit the PRACK request when it receives a retransmission of the provisional response being acknowledged, although doing so does not create a protocol error. Once a reliable provisional response is received, retransmissions of that response MUST be discarded. A response is a retransmission when its dialog ID, CSeq and RSeq match the original response. The UAC MUST maintain a sequence number which indicates the most recently received in-order reliable provisional response for the initial request. This sequence number MUST be maintained until a final response is received for the initial request. Its value MUST be initialized to the RSeq header in the first reliable provisional response received for the initial request. Handling of subsequent reliable provisional responses for the same initial request follows the same rules as above, with the following difference. Reliable provisional responses are guaranteed to be in order. As a result, if the UAC receives another reliable provisional response to the same request, and its RSeq value isn't one higher than the value of the sequence number, that response MUST NOT be acknowledged with a PRACK, and MUST NOT be processed further by the TU. An implementation MAY discard the response, or MAY cache the response in the hopes of receiving the missing responses. The UAC MAY acknowledge reliable provisional responses received after the final response, or MAY discard them. 19 Transport The transport layer is responsible for the actual transmission of requests and responses over network transports. This includes determination of the connection to use for a request or response, in the case of connection oriented transports. The transport layer is responsible for managing any persistent connections (for transports like TCP, TLS and SCTP) including ones it opened, as well as ones opened to it. This includes connections Various Authors [Page 131] Internet Draft SIP January 28, 2002 opened by the client or server transports, so that connections are shared between client and server transport functions. These connections are indexed by the [address, port, transport] at the far end of the connection. When a connection is opened by the transport layer, this index is set to the destination IP, port and transport. When the connection is accepted by the transport layer, this index is set to the source IP, port and transport. Note that, because the source port is often ephemeral, connections accepted by the transport layer will frequently not be reused. The result is that two proxies in a "peering" relationship using a connection oriented transport will frequently have two connections in use, one for transactions initiated in each direction. It is RECOMMENDED that connections be kept open for some implementation defined duration after the last message was sent or received over that connection. This duration SHOULD at least equal the longest amount of time the element would need in order to bring a transaction from instantiation to the terminated state. This is to insure that transactions complete over the same connection they are initiated on (i.e., request, response, and in the case of INVITE, ACK for non-2xx responses)). This usually means at least the maximum of T3 and 64*T1. However, it could be larger in an element that has a TU that is using a large value for timer C, for example. All SIP elements MUST implement UDP and TCP. Other transports MAY be implemented by any entity. Making TCP mandatory for UA is a substantial change from RFC 2543. It has arisen out of the need to handle larger messages, which MUST use TCP, as discussed below. Thus, even if an element never sends large messages, it may receive one, and needs to be able to do that. 19.1 Clients 19.1.1 Sending Requests The client side of the transport layer is responsible for sending the request and receiving responses. The user of the transport layer passes the client transport the request, an IP address, port, transport, and possibly TTL for multicast destinations. If a request is within 500 bytes of the path MTU, or if it is larger than 1000 bytes when the path MTU is unknown, it MUST be sent using TCP. This is to prevent fragmentation of messages over UDP, and to provide congestion control for larger messages. However, implementations MUST be able to handle messages up to the maximum Various Authors [Page 132] Internet Draft SIP January 28, 2002 datagram packet size. For UDP, this size is 65,535 bytes, including headers. The 500 byte "buffer" between the message size and the MTU accomodates the fact that the response in SIP can be larger than the request. This happens due to the addition of Record-Route headers to the responses to INVITE, for example. With the extra buffer, the response can be 500 bytes larger than the request, and still not be fragmented. 1000 is chosen when path MTU is not known, based on the assumption of a 1500 byte ethernet MTU A client that sends a request to a multicast address MUST add the "maddr" parameter to its Via header field, and SHOULD add the "ttl" parameter. (In that case, the maddr parameter SHOULD contain the destination multicast address, although under exceptional circumstances it MAY contain a unicast address.) Requests sent to multicast groups SHOULD be scoped to ensure that they are not forwarded beyond the administrative domain to which they were targeted. This scoping MAY be done with either TTL or administrative scopes [17], depending on what is implemented in the network. It is important to note that the layers above the transport layer do not operate differently for multicast as opposed to unicast requests. This means that SIP treats multicast more like anycast, assuming that there is a single recipient generating responses to requests. If this is not the case, the first response will end up "winning", based on the client transaction rules. Any other responses from different UA will appear as retransmissions and be discarded. This limits the utility of multicast to cases where an anycast type of function is desired, such as registrations. Before a request is sent, the client transport MUST insert a value of the sent-by field into the Via header. This field contains an IP address or host name, and port. The usage of an FQDN is RECOMMENDED. This field is used for sending responses under certain conditions. For reliable transports, the response is normally sent on the connection the request was received on. Therefore, the client transport MUST be prepared to receive the response on the same connection used to send the request. Under error conditions, the server may attempt to open a new connection to send the response. To handle this case, the transport layer MUST also be prepared to receive an incoming connection on the source IP address that the request was sent from, and port number in the sent-by field. It also MUST be prepared to receiving incoming connections on any address and port which would be selected by a server based on the procedures Various Authors [Page 133] Internet Draft SIP January 28, 2002 described in Section 5 of [8]. For unreliable unicast transports, the client transport MUST be prepared to receive responses on the source IP address that the request is sent from (as responses are sent back to the source address), but the port number in the sent-by field. Furthermore, as with reliable transports, in certain cases the response will be sent elsewhere. The client MUST be prepared to receive responses on any address and port which would be selected by a server based on the procedures described in Section 5 of [8]. For multicast, the client transport MUST be prepared to receive responses on the same multicast group and port that the request is sent to (e.g., it needs to be a member of the multicast group it sent the request to.) If a request is destined to an IP address, port, and transport to which an existing connection is open, it is RECOMMENDED that this connection be used to send the request, but another connection MAY be opened and used. If a request is sent using multicast, it is sent to the group address, port, and TTL provided by the transport user. If a request is sent using unicast unreliable transports, it is sent to the IP address and port provided by the transport user. 19.1.2 Receiving Responses When a response is received, the client transport examines the top Via header. If the value of the sent-by parameter in that header does not correspond to a value that the client transport is configured to insert into requests, the response MUST be rejected. If there are any client transactions in existence, the client transport uses the matching procedures of Section 17.1.3 to attempt to match the response to an existing transaction. If there is a match, the response MUST be passed to that transaction. Otherwise, the response MUST be passed to the core (whether it be stateless proxy, stateful proxy, or UA) for further processing. Handling of these "stray" responses is dependent on the core (a stateless proxy will forward all responses, for example). 19.2 Servers 19.2.1 Receiving Requests When the server transport receives a request over any transport, it MUST examine the value of the sent-by parameter in the top Via header Various Authors [Page 134] Internet Draft SIP January 28, 2002 field. If the host portion of the sent-by parameter contains a domain name, or if it contains an IP address that differs from the packet source address, the server MUST add a "received" attribute to that Via header field. This attribute MUST contain the source address that the packet was received from. This is to assist the server transport layer in sending the response, since it must be sent to the source IP address that the request came from. Consider a request received by the server transport which looks like, in part: INVITE sip:bob@Biloxi.com SIP/2.0 Via: SIP/2.0/UDP bobspc.biloxi.com:5060 The request is received with a source IP address of 1.2.3.4. Before passing the request up, the transport would add a received parameter, so that the request would look like, in part: INVITE sip:bob@Biloxi.com SIP/2.0 Via: SIP/2.0/UDP bobspc.biloxi.com:5060;received=1.2.3.4 Next, the server transport attempts to match the request to the server transaction. It does so using the matching rules described in Section 17.2.3. If a matching server transaction is found, the request is passed to that transaction for processing. If no match is found, the request is passed to the core, which may decide to construct a new server transaction for that request. Note that when a UAS core sends a 2xx response to INVITE, the server transaction is destroyed. This means that when the ACK arrives, there will be no matching server transaction, and based on this rule, the ACK is passed to the UAS core, where it is processed. 19.2.2 Sending Responses The server transport uses the value of the top Via header in order to determine where to send a response. It MUST follow the following process: o If the "sent-protocol" is a reliable transport protocol such as TCP, TLS or SCTP, the response MUST be sent using the existing connection to the source of the original request that created the transaction, if that connection is still open. Various Authors [Page 135] Internet Draft SIP January 28, 2002 This does require the server transport to maintain an association between server transactions and transport connections. If that connection is no longer open, the server MAY open a connection to the IP address in the received parameter, if present, using the port in the sent-by value, or the default port for that transport, if no port is specified (5060 for UDP and TCP, 5061 for TLS and SSL). If that connection attempt fails, the server SHOULD use the procedures in [8] for servers in order to determine the IP address and port to open the connection and send the response to. o Otherwise, if the Via header field contains a "maddr" parameter, forward the response to the address listed there, using the port indicated in "sent-by", or port 5060 if none is present. If the address is a multicast address, the response SHOULD be sent using the TTL indicated in the "ttl" parameter, or with a TTL of 1 if that parameter is not present. o Otherwise (for unreliable unicast transports), if the top Via has a received parameter, send the response to the address in the "received" parameter, using the port indicated in the "sent-by" value, or using port 5060 if none is specified explicitly. If this fails, e.g., elicits an ICMP "port unreachable" response, send the response to the address in the "sent-by" parameter. The address to send to is determined by following the procedures defined in Section 5 of [8]. o Otherwise, if it is not receiver-tagged, send the response to the address indicated by the "sent-by" value, using the procedures in Section 5 of [8]. 19.3 Framing In the case of message oriented transports (such as UDP), if the message has a Content-Length header, the message body is assumed to contain that many bytes. If there are additional bytes in the transport packet below the end of the body, they MUST be discarded. If the transport packet ends before the end of the message body, this is considered an error. If the message is a response, it MUST be discarded. If its a request, the element SHOULD generate a 400 class response. If the message has no Content-Length header, the message body is assumed to end at the end of the transport packet. In the case of stream oriented transports (such as TCP), the Content-Length header indicates the size of the body. The Content- Length header MUST be used with stream oriented transports. 19.4 Error Handling Various Authors [Page 136] Internet Draft SIP January 28, 2002 Error handling is independent of whether the message was a request or response. If the transport user asks for a message to be sent over an unreliable transport, and the result is an ICMP error, the behavior depends on the type of ICMP error. A host, network, port or protocol unreachable errors, or parameter problem errors SHOULD cause the transport layer to inform the transport user of a failure in sending. Source quench and TTL exceeded ICMP errors SHOULD be ignored. If the transport user asks for a request to be sent over a reliable transport, and the result is a connection failure, the transport layer SHOULD inform the transport user of a failure in sending. 20 Usage of HTTP Authentication SIP provides a stateless challenged-based mechanism for authentication that is based on authentication in HTTP. Any time that a proxy server or user agent receives a request (with the exceptions given in Section 20.1), it MAY challenge the initiator of the request to provide assurance of its identity. Once the originator has been identified, the recipient of the request SHOULD ascertain whether or not this user is authorized to make the request in question. No authorization systems are recommended or discussed in this document. The "Digest" authentication mechanism described in this section provides message authentication and replay protection only, without message integrity or confidentiality. Protective measures above and beyond those provided by Digest need to be taken to prevent active attackers from modifying SIP requests and responses. Note that due to its weak security, the usage of "Basic" authentication has been deprecated. Servers MUST NOT accept credentials using the "Basic" authorization scheme, and servers also MUST NOT challenge with "Basic". This is a change from RFC 2543. 20.1 Framework The framework for SIP authentication closely parallels that of HTTP (RFC 2617 [23]). In particular, the BNF for auth- scheme, auth-param, challenge, realm, realm-value, and credentials is identical (although the usage of "Basic" as a scheme is not permitted). The 401 (Unauthorized) response is used by user agent servers in SIP to challenge the identity of a user agent client. Additionally, registrars and redirect servers MAY make use of 401 (Unauthorized) responses for authentication, but proxies MUST NOT, and instead MAY use the 407 (Proxy Authentication Required) response. The Various Authors [Page 137] Internet Draft SIP January 28, 2002 requirements for inclusion of the Proxy-Authenticate, Proxy- Authorization, WWW-Authenticate, and Authorization in the various messages are identical to those described in RFC 2617 [23]. Since SIP does not have the concept of a canonical root URL, the notion of protection spaces is interpreted differently in SIP. The realm string alone defines the protection domain. This is a change from RFC 2543, in which the Request-URI and the realm together defined the protection domain; this definition gave rise to some amount of confusion since the Request-URI sent by the UAC and the Request-URI received by the server issuing a challenge might be different, and indeed the final form of the Request-URI might not be known to the UAC. Also, the previous definition depended on the presence of a SIP URI in the Request-URI, and seemed to rule out alternative URI schemes (like for example the tel URL). Operators of user agents or proxy servers that will authenticate received requests MUST adhere to the following guidelines for creation of a realm string for their server: o Realm strings MUST be globally unique. It is RECOMMENDED that a realm string contain a hostname or domain name, following the recommendation in Section 3.2.1 of RFC 2617 [[23]]. o Realm strings SHOULD present a human-readable identifier that can be rendered to a user. For example: INVITE sip:bob@biloxi.com SIP/2.0 WWW-Authenticate: Digest realm="biloxi.com", <...> Generally, SIP authentication is meaningful for a specific realm, a protection domain. Thus, for Digest authentication, each such protection domain has its own set of user names and secrets. If a server does not care about authenticating individual users, it may make sense to establish a "global" user name and secret for its realm as a default challenge if a particular Request-URI does not have its own realm or set of user names, For example, an INVITE to gateways, MAY have their own device-specific credentials for particular realms. While a server can legitimately challenge most SIP requests, there are two requests defined by the SIP standard today that require special handling for authentication: ACK and CANCEL. Various Authors [Page 138] Internet Draft SIP January 28, 2002 Complications of the ACK method arise because it requires no response. Under an authentication scheme that uses responses to carry values used to compute nonces (such as Digest), some problems come up for any requests that take no response (including ACK). For this reason any credentials in the INVITE that were accepted by a server MUST be accepted by that server for the ACK. UACs creating an ACK message should duplicate all of the Authorization and Proxy- Authorization headers that appeared in the INVITE to which the ACK corresponds. Servers MUST NOT attempt to challenge an ACK. Although the CANCEL method does take a response (a 2xx), servers MUST NOT attempt to challenge CANCEL requests since these requests cannot be resubmitted. Generally, a CANCEL request SHOULD be accepted by a server if it comes from the same host that sent the request being cancelled (provided that some sort of transport or network layer security association, as described in Section 22.2.1, is in place). When a challenge is received by a UAC, it SHOULD render to the user the contents of the "realm" parameter in the challenge (which appears in either a WWW-Authenticate header or Proxy-Authenticate header) if the UAC device does not already know of a credential for the realm in question. A service provider that pre-configures UAs with credentials for its realm should be aware that users will not have the opportunity to present their own credentials for this realm when challenged at a pre-configured device. Finally, note that even if a UAC can locate credentials that are associated with the proper realm, there is always a potential that these credentials may no longer be valid, or that for whatever reason the challenging server will not accept these credentials. In this instance a server will commonly repeat its challenge. A UAC MUST NOT reattempt requests with the credentials that have just been rejected (unless the request was rejected because of a stale nonce). 20.2 User-to-User Authentication When a UAS receives a request from a UAC, the UAS MAY authenticate the originator before the request is processed. If no credentials (in the Authorization header field) are provided in the request, the UAS can challenge the originator to provide credentials by rejecting the request with a 401 (Unauthorized) status code. The WWW-Authenticate response-header field MUST be included in 401 (Unauthorized) response messages. The field value consists of at least one challenge that indicates the authentication scheme(s) and parameters applicable to the Request-URI. See [H14.47] for a definition of the syntax. Various Authors [Page 139] Internet Draft SIP January 28, 2002 An example of the WWW-Authenticate header field in a 401 challenge is: WWW-Authenticate: Digest realm="biloxi.com", qop="auth,auth-int", nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093", opaque="5ccc069c403ebaf9f0171e9517f40e41" When the originating UAC receives the 401 (Unauthorized), it SHOULD, if it is able, re-originate the request with the proper credentials. The UAC may require input from the originating user before proceeding. Once authentication credentials have been supplied (either directly by the user, or discovered in an internal keyring), user agents SHOULD cache the credentials for a given value of the To header and "realm" and attempt to re-use these values on the next request for that destination. UAs MAY cache credentials in any way they would like. Once credentials have been located, any user agent that wishes to authenticate itself with a UAS or registrar -- usually, but not necessarily, after receiving a 401 (Unauthorized) response -- MAY do so by including an Authorization header field with the request. The Authorization field value consists of credentials containing the authentication information of the user agent for the realm of the resource being requested as well as parameters required in support of authentication and replay protection. An example of the Authorization header is: Authorization: Digest username="bob", realm="biloxi.com", nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093", uri=sip:alice@atlanta.com, qop=auth, nc=00000001, cnonce="0a4f113b", response="6629fae49393a05397450978507c4ef1", opaque="5ccc069c403ebaf9f0171e9517f40e41" Various Authors [Page 140] Internet Draft SIP January 28, 2002 When a UAC resubmits a request with its credentials after receiving a 401 (Unauthorized) or 407 (Proxy Authentication Required) response, it MUST increment the CSeq header field as it would normally when sending an updated request. 20.3 Proxy to User Authentication Similarly, when a UAC sends a request to a proxy server, the proxy server MAY authenticate the originator before the request is processed. If no credentials (in the Proxy-Authorization header field) are provided in the request, the UAS can challenge the originator to provide credentials by rejecting the request with a 407 (Proxy Authentication Required) status code. The proxy MUST populate the 407 (Proxy Authentication Required) message with a Proxy- Authenticate header applicable to the proxy for the requested resource. The use of the Proxy-Authentication and Proxy-Authorization parallel that described in [23], with one difference. Proxies MUST NOT add the Proxy-Authorization header. 407 (Proxy Authentication Required) responses MUST be forwarded upstream towards the UAC following the procedures for any other response. It is the client's responsibility to add the Proxy-Authorization header containing credentials for the realm of the proxy which has asked for authentication. If a proxy were to resubmit a request with a Proxy- Authorization header field, it would need to increment the CSeq in the new request. However, this would mean that the UAC which submitted the original request would discard a response from the UAS, as the CSeq value would be different. When the originating UAC receives the 407 (Proxy Authentication Required) it SHOULD, if it is able, re-originate the request with the proper credentials. It should follow the same procedures for the display of the "realm" parameter that are given above for responding to 401. The UAC SHOULD also cache the credentials used in the re- originated request. The following rule is RECOMMENDED for proxy credential caching: If a UA receives a Proxy-Authenticate header in a 401/407 response to a request with a particular Call-ID, it should incorporate credentials for that realm in all subsequent requests that contain the same Call-ID. These credentials MUST NOT be cached across dialogs; however, if a UA is configured with the realm of its local outbound proxy, when one exists, then the UA MAY cache credentials Various Authors [Page 141] Internet Draft SIP January 28, 2002 for that realm across dialogs. Note that this does mean a future requests in a dialog could contain credentials that are not needed by any proxy along the Route header path. Any user agent that wishes to authenticate itself to a proxy server -- usually, but not necessarily, after receiving a 407 (Proxy Authentication Required) response -- MAY do so by including a Proxy- Authorization header field with the request. The Proxy-Authorization request-header field allows the client to identify itself (or its user) to a proxy which requires authentication. The Proxy- Authorization header field value consists of credentials containing the authentication information of the user agent for the proxy and/or realm of the resource being requested. A Proxy-Authorization header field applies only to the proxy whose realm is identifier in the "realm" parameter (this proxy may previously have demanded authentication using the Proxy-Authenticate field). When multiple proxies are used in a chain, the Proxy- Authorization header field MUST NOT be consumed by any proxy whose realm does not match the "realm" parameter specified in the Proxy- Authorization header. Note that if an authentication scheme is used in the Proxy- Authorization that does not support realms, a proxy server MUST attempt to parse all Proxy-Authorization headers to determine whether or not one of them has what it considers to be valid credentials. Because this is potentially very time consuming in large networks, proxy servers SHOULD use an authentication scheme that supports realms in the Proxy-Authorization header. If a request is forked (as described in Section 16.6, various proxy servers and/or user agents may wish to challenge the UAC. In this case the forking proxy server is responsible for aggregating these challenges into a single response. Each WWW-Authenticate and Proxy- Authenticate received in responses to the forked request MUST be placed into the single response that is sent by the forking proxy to the user agent; the ordering of these headers is not significant. When a proxy server issues a challenge in response to a request, it will not proxy the request until the UAC has provided valid credentials. A forking proxy may forward a request simultaneously to multiple proxy servers that require authentication, each of which in turn will not forward the request until the originating UAC has authenticated itself in their respective realm. If the UAC does not provide credentials for each of these challenges, then the proxy servers that issued the challenges will not Various Authors [Page 142] Internet Draft SIP January 28, 2002 forward requests to user agents where the destination user might be located, and therefore, the virtues of forking are largely lost. If at least one UAS responds to a forked request with a challenge, than a 401 (Unauthorized) MUST be sent as the aggregated response by the forking proxy to the UAC; otherwise, if only proxy servers respond, a 407 MUST be used. When resubmitting its request in response to a 401 (Unauthorized) or 407 (Proxy Authentication Required) that contains multiple challenges, a UAC MAY include an Authorization for each WWW- Authenticate and Proxy-Authorization for each Proxy-Authenticate for which the UAC wishes to supply a credential. As noted above, multiple credentials in a request SHOULD be differentiated by the "realm" parameter. It is possible for multiple challenges associated with the same realm to appear in the same 401 (Unauthorized) or 407 (Proxy Authentication Required). This can occur, for example, when multiple proxies within the same administrative domain, which use a common realm, are reached by a forking request. See [H14.34] for a definition of the syntax of Proxy- Authentication and Proxy-Authorization. 20.4 The Digest Authentication Scheme This section describes the modifications and clarifications required to apply the HTTP Digest authentication scheme to SIP. The SIP scheme usage is almost completely identical to that for HTTP [23]. Since RFC 2543 is based on HTTP Digest as defined in RFC 2069 [24], SIP servers supporting RFC 2617 MUST ensure they are backwards compatible with RFC 2069. Procedures for this backwards compatibility are specified in RFC 2617. Note however that servers MUST NOT accept or request Basic authentication. 20.4.1 HTTP Digest The rules for Digest authentication follow those defined in [23], with "HTTP 1.1" replaced by "SIP/2.0" in addition to the following differences: 1. The URI included in the challenge has the following BNF: URI = SIP-URI Various Authors [Page 143] Internet Draft SIP January 28, 2002 2. The BNF in RFC 2617 has an error in that the 'uri' parameter of the Authorization header for HTTP Digest authentication is not enclosed in quotation marks. (The example in Section 3.5 of RFC 2617 is correct.) For SIP, the 'uri' MUST be enclosed in quotation marks. 3. The BNF for digest-uri-value is: digest-uri-value = Request-URI ; as defined in Section 27 4. The example procedure for choosing a nonce based on Etag does not work for SIP. 5. The text in RFC 2617 [23] regarding cache operation does not apply to SIP. 6. RFC 2617 [23] requires that a server check that the URI in the request line, and the URI included in the Authorization header, point to the same resource. In a SIP context, these two URI's may actually refer to different users, due to forwarding at some proxy. Therefore, in SIP, a server MAY check that the Request-URI in the Authorization header corresponds to a user for whom that the server is willing to accept forwarded or direct requests. 7. As a clarification to the calculation of the A2 value for message integrity assurance in the Digest authentication scheme, implementers should assume, when the entity-body is empty (i.e. when SIP messages have no body) that the hash of the entity-body resolves to the MD5 hash of an empty string, or: H(entity-body) = MD5("") = "d41d8cd98f00b204e9800998ecf8427e" 8. RFC 2617 notes that a cnonce value MUST NOT be sent in an Authorization (and by extension Proxy-Authorization) header if no qop directive as been sent. Therefore, any algorithms that have a dependency on the cnonce (including "MD5-Sess") require that the qop directive be sent. Use of the "qop" parameter is optional in RFC 2617 for the purposes of backwards compatibility with RFC 2069; since RFC 2543 was based on RFC 2069, the "qop" parameter must unfortunately Various Authors [Page 144] Internet Draft SIP January 28, 2002 remain optional for clients and servers to receive. However, servers MUST always send a "qop" parameter in WWW-Authenticate and Proxy-Authenticate headers. If a client receives a "qop" parameter in a challenge header, it MUST send the "qop" parameter in any resulting authorization header. RFC 2543 did not allow usage of the Authentication-Info header (it effectively used RFC 2069). However, we now allow usage of this header, since it provides integrity checks over the bodies and provides mutual authentication. RFC 2617 [23] defines mechanisms for backwards compatibility using the qop attribute in the request. These mechanisms MUST be used by a server to determine if the client supports the new mechanisms in RFC 2617 that were not specified in RFC 2069. 21 S/MIME SIP messages carry MIME bodies and the MIME standard includes mechanisms for securing MIME contents to ensure both integrity and confidentiality (including the 'multipart/signed/' and 2630 [26] and RFC 2633 [27]). Implementers should note, however, that there may be rare network intermediaries (not typical proxy servers) that rely on viewing or modifying the bodies of SIP messages (especially SDP), and that secure MIME may prevent these sorts of intermediaries from functioning. This applies particularly to certain types of firewalls. Note that the PGP mechanism for encrypting the headers and bodies of SIP messages described in RFC 2543 has been deprecated. 21.1 S/MIME Certificates The certificates that are used to identify an end-user for the purposes of S/MIME differ from those used by servers in one important respect - rather than asserting that the identity of the holder corresponds to a particular hostname, these certificates assert that the holder is identified by an end-user address - this address is composed of the concatenation of the "userinfo" "@" and "domainname" portions of a SIP URI (in other words, an email address of the form "bob@biloxi.com"), most commonly corresponding to a user's address of record. These certificates are used to sign or encrypt bodies of SIP messages. Bodies are signed with the private key of the sender (who Various Authors [Page 145] Internet Draft SIP January 28, 2002 may include their public key with the message as appropriate), but bodies are encrypted with the public key of the intended recipient. Obviously, senders must have foreknowledge of the public key of recipients in order to encrypt message bodies. Public keys can be stored within a user agent on a virtual keyring. Each user agent that supports S/MIME MUST contain a keyring specifically for end-users certificates. This keyring should map between addresses of record and corresponding certificates, including any associated with the owner or operator of the user agent, when appropriate. Over time, users SHOULD use the same certificate when they populate the originating URI of signaling (the From header) with the same address of record. Any mechanisms that depend on the existence of end-user certificates, however, have a serious limitation in that there is virtually no consolidated authority today that provides certificates for end-user applications. But if at all possible, users SHOULD acquire certificates from known public certificate authorities. As an alternative, users MAY create self-signed certificates. The implications of self-signed certificates are explored further in Section 22.4.2. Above and beyond the problem of acquiring an end-user certificate, there are few well-known centralized directories that distribute end-user certificates. However, the holder of a certificate SHOULD publish their certificate in any public directories as appropriate. Similarly, UACs SHOULD support a mechanism for importing (manually or automatically) certificates discovered in public directories corresponding to the target URIs of SIP requests. 21.2 S/MIME Key Exchange SIP itself can also be used as a means to distribute public keys in the following manner. Whenever the CMS SignedData message is used in S/MIME for SIP, it MUST contain the certificate bearing the public key necessary to verify the signature. When a UAC sends a request containing an S/MIME body that initiates a dialog, or sends a non-INVITE request outside the context of a dialog, the UAC SHOULD structure the body as an S/MIME EnvelopedData, the UAC should send the EnvelopedData message encapsulated within a SignedData message. When a UAS receives a request containing an S/MIME CMS body which includes a certificate, the UAS SHOULD first verify the certificate, Various Authors [Page 146] Internet Draft SIP January 28, 2002 if possible, with any available certificate authority. The UAS SHOULD also determine the subject of the certificate and compare this value to the From field of the request. If the certificate cannot be verified, because it is self-signed, or signed by no known authority, the UAS SHOULD notify the user of the status of the certificate (including the subject of the certificate, its signator, and any key fingerprint information) and request explicit permission before proceeding. If the certificate was successfully verified and the subject of the certificate corresponds to the From header field of the SIP request, or if the user (after notification) explicitly authorizes the use of the certificate, the UAS SHOULD add this certificate to a local keyring, indexed by the address of record of the holder of the certificate. When a UAS sends a response containing an S/MIME body that answers the first request in a dialog, or a response to a non-INVITE request outside the context of a dialog, the UAS SHOULD structure the body as a S/MIME 'multipart/signed' CMS SignedData body; if the desired CMS service is EnvelopedData, the UAS SHOULD send the EnvelopedData message encapsulated within a SignedData message. If the S/MIME body received by the UAS was encrypted with a public key recognized by the UAS, it MAY opt not to sign its response when appropriate. When a UAC receives a response containing an S/MIME CMS body which includes a certificate, the UAC SHOULD first verify the certificate, if possible, with any available certificate authority. The UAC SHOULD also determine the subject of the certificate and compare this value to the To field of the response; although the two may very well be different, and this is not necessarily indicative of a security breach. If the certificate cannot be verified, because it is self- signed, or signed by no known authority, the UAC SHOULD notify the user of the status of the certificate (including the subject of the certificate, its signator, and any key fingerprint information) and request explicit permission before proceeding. If the certificate was successfully verified and the subject of the certificate corresponds to the To header in the response, or if the user (after notification) explicitly authorizes the use of the certificate, the UAC SHOULD add this certificate to a local keyring, indexed by the address of record of the holder of the certificate. If the UAC had not transmitted its own certificate to the UAS in any previous transaction, it SHOULD use a CMS SignedData body for its next request or response. On future occasions, when the UA receives requests or responses that contain a From header field corresponding to a value in its keyring, the UA SHOULD compare the certificate offered in these messages with the existing certificate in its keyring. If there is a discrepancy, the UA SHOULD notify the user of a change of the certificate (preferably in terms that indicate that this is a potential security Various Authors [Page 147] Internet Draft SIP January 28, 2002 breach) and acquire the user's permission before continuing to process the signaling. If the user authorizes this certificate, it MUST be added to the keyring alongside any previous value(s) for this address of record. Note well however, that this key exchange mechanism does not guarantee the secure exchange of keys when self-signed certificates, or certificates signed by an obscure authority, are used - it is vulnerable to well-known attacks. In the opinion of the authors, however, the security it provides is proverbially better than nothing; it is in fact comparable to the widely used SSH application. These limitations are explored in greater detail in Section 22.4.2. If a user agent receives an S/MIME body that has been encrypted with a public key unknown to the recipient, it MUST reject the request with a 493 (Undecipherable) response. This response SHOULD contain a valid certificate for the respondent (corresponding, if possible, to any address of record given in the To header of the rejected request) within a MIME body. A 493 (Undecipherable) sent without any certificate indicates that the respondent cannot or will not utilize S/MIME. Finally, if during the course of a dialog a user agent receives a certificate in a CMS SignedData message that does not correspond with the certificates previously exchanged during a dialog, the user agent MUST notify its user of the change, preferably in terms that indicate that this is a potential security breach. 21.3 Securing MIME bodies There are two types of secure MIME bodies that are of interest to SIP: use of these bodies should follow the S/MIME specification ([27]) with a few variations. o signatures. This allows backwards compatibility with non-S/MIME- compliant recipients. o If a UAC has no certificate on its keyring associated with the address of record to which it wants to send a request, it cannot send an encrypted 'application/pkcs7-mime' MIME message. UACs MAY send an initial request such as an OPTIONS message with a CMS detached signature in order to solicit the certificate of the remote side (the signature SHOULD be over a 'message/sip' body of the type described in Section 21.4). Various Authors [Page 148] Internet Draft SIP January 28, 2002 o Senders of S/MIME bodies SHOULD use the 'SMIMECapabilities' (see Section 2.5.2 of [27]) attribute to express their capabilities and preferences for further communications. Note especially that senders MAY use the 'preferSignedData' capability to encourage receivers to respond with CMS SignedData messages (for example, when sending an OPTIONS request as described above). o S/MIME implementations MUST at a minimum support SHA1 as a digital signature algorithm, and 3DES as an encryption algorithm; all other signature and encryption algorithms MAY be supported. Implementations can negotiate support for these algorithms with the 21.4 Tunneling SIP in MIME As a means of providing some degree of end-to-end authentication, integrity or confidentiality for SIP headers, S/MIME can encapsulate entire SIP messages within MIME bodies of type "message/sip" and then apply MIME security to these bodies in the same manner employed for typical SIP bodies. Note that these "message/sip" bodies can be sent as a part of a MIME "multipart/mixed" body if another MIME types (such as SDP) should also be used in the request. 21.4.1 Tunneling Integrity and Authentication Tunneling SIP messages within S/MIME bodies can provide integrity for SIP headers if the headers which the sender wishes to secure are replicated in a "message/sip" MIME body signed with a CMS detached signature. Provided that the "message/sip" body contains at least the fundamental dialog identifiers (To, From, Call-ID, CSeq), then a signed MIME body can provide limited authentication. At the very least, if the certificate used to sign the body is unknown to the recipient and cannot be verified, the signature can be used to ascertain that a later request in a dialog was transmitted by the same certificate-holder that initiated the dialog. If the recipient of the signed MIME body has some stronger incentive to trust the certificate (they were able to verify it, acquire it from a trusted repository, or they've used it frequently) then the signature can be taken as a stronger assertion of the identity of the subject of the certificate. In order to eliminate possible confusions about the addition or subtraction of entire headers, senders SHOULD replicate all headers Various Authors [Page 149] Internet Draft SIP January 28, 2002 from the request within the signed body. Any message bodies that require integrity protection SHOULD be attached to the "inner" message. Upon receipt of a SIP message with a signed "message/sip" body, recipients may compare headers in the "outer" message with headers in the "inner" message. At the discretion of the recipient, if significant discrepancies between the two exist, the message MAY be rejected with a 403 (Forbidden) response if it is a request, or any existing dialog MAY be terminated if a security violation has occurred. User agents SHOULD notify users of this circumstance and request explicit guidance on how to proceed. Provided that the signature is valid for the "inner" message, headers in the inner message SHOULD be preferred to headers in the "outer" message. Many SIP headers are altered of necessity as messages are routed through proxy servers. These include, but are not necessarily limited to, the Request-URI, Via headers, Record-Route and Route headers, the Max-Forwards header, and the Proxy-Authorization header; note that extensions to SIP, or nonstandard (X-) headers, may also result in headers that are added or subtracted from messages as they traverse the network. A variation in these headers SHOULD NOT be interpreted as a breach of integrity by the recipient of a signed message. The following is an example of the use of a tunneled "message/sip" body: INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: Content-Type: multipart/signed; protocol="application/pkcs7-signature"; micalg=sha1; boundary=boundary42 --boundary42 Content-Type: message/sip INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 Various Authors [Page 150] Internet Draft SIP January 28, 2002 CSeq: 314159 INVITE Contact: Content-Type: application/sdp Content-Length: 147 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com s=Session SDP c=IN IP4 pc33.atlanta.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000 --boundary42 Content-Type: application/pkcs7-signature; name=smime.p7s Content-Transfer-Encoding: base64 Content-Disposition: attachment; filename=smime.p7s ghyHhHUujhJhjH77n8HHGTrfvbnj756tbB9HG4VQpfyF467GhIGfHfYT6 4VQpfyF467GhIGfHfYT6jH77n8HHGghyHhHUujhJh756tbB9HGTrfvbnj n8HHGTrfvhJhjH776tbB9HG4VQbnj7567GhIGfHfYT6ghyHhHUujpfyF4 7GhIGfHfYT64VQbnj756 --boundary42- 21.4.2 Tunneling Encryption It may also be desirable to use this mechanism to encrypt a "message/sip" MIME body within a CMS EnvelopedData message S/MIME body, but in practice, most headers are of at least some use to the network; the general use of encryption with S/MIME is to secure message bodies like SDP rather than message headers. Some informational headers, such as the Subject or Organization could perhaps warrant end-to-end security. Headers defined by future SIP applications might also require obfuscation. Another possible application of encrypting headers is selective anonymity. A request could be constructed with a From header field that contains no personal information (e.g., sip:anonymous@anonymizer.com). However, a second From header field containing the genuine address of record of the originator could be encrypted within a "message/sip" MIME body where it will only be visible to the endpoints of a dialog. In the following example, the text boxed in asterisks ("*") is encrypted: Various Authors [Page 151] Internet Draft SIP January 28, 2002 INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: Content-Type: application/pkcs7-mime; smime-type=enveloped-data; name=smime.p7m Content-Transfer-Encoding: base64 Content-Disposition: attachment; filename=smime.p7m ******************************************************* * Content-Type: application/sdp * * * * v=0 * * o=alice 53655765 2353687637 IN IP4 pc33.atlanta.com * * s=- * * t=0 0 * * c=IN IP4 pc33.atlanta.com * * m=audio 3456 RTP/AVP 0 1 3 99 * * a=rtpmap:0 PCMU/8000 * ******************************************************* 22 Security Considerations SIP is not an easy protocol to secure. Its use of intermediaries, its multi-faceted trust relationships, its expected usage between elements with no trust at all, and its user-to-user operation make security far from trivial. Security solutions are needed that are deployable today, without extensive coordination, in a wide variety of environments and usages. In order to meet these diverse needs, several distinct mechanisms applicable to different aspects and usages of SIP will be required. Note that the security of SIP signaling itself has no bearing on the security of protocols used in concert with SIP such as RTP, or with the security implications of any specific bodies SIP might carry (although MIME security plays a substantial role in securing SIP). Any media associated with a session can be encrypted end-to-end independently of any associated SIP signaling. Media encryption is outside the scope of this document. The considerations that follow first examine a set of classic threat models which broadly identify the security needs of the SIP protocol. Various Authors [Page 152] Internet Draft SIP January 28, 2002 The set of security services required to address these threats is then detailed, followed by an explanation of several security mechanisms that can be used to provide these services. Next, the requirements for implementers of SIP are enumerated, along with exemplary deployments in which these security mechanisms could be used to improve the security of SIP. Some notes on privacy conclude this section. 22.1 Threat Models This section details some threats that should be common to most deployments of SIP. These threats have been chosen specifically to illustrate each of the security services that SIP requires. The following examples by no means provide an exhaustive list of the threats against the SIP protocol; rather, these are "classic" threats that demonstrate the need for particular security services which can potentially prevent whole categories of threats. 22.1.1 Registration Hijacking The SIP registration mechanism allows a user agent to identify itself to a registrar as a device at which a user (designated by an address of record) is located. A registrar assesses the identity asserted in the From header field of a REGISTER message to determine whether or not this request can modify the contact addresses associated with the address of record in the To header field; while these two fields are frequently the same, there are many valid deployments in which a third-party may register contacts on a user's behalf. The From header of a SIP request, however, can essentially be modified arbitrarily by the owner of a user agent, and this opens the door to malicious registrations. An attacker that successfully impersonates a party authorized to change contacts associated with an address of record could, for example, de-register all existing contacts for a URI and then register their own device as the appropriate contact address, thereby directing all requests for the affected user to the attacker's device. This threat belongs to a family of threats that rely on the absence of cryptographic assurance of a request's originator. Any SIP UAS that represents a valuable service (a gateway that interworks SIP requests with traditional telephone calls, for example) might want to control access to its resources by authenticating requests that it receives. Even end-user UAs, for example SIP phones, have an interest in ascertaining the identities of originators of requests. This threat demonstrates the need for security services that enable Various Authors [Page 153] Internet Draft SIP January 28, 2002 SIP entities to authenticate the originators of requests. 22.1.2 Impersonating a Server The domain to which a request is destined is generally specified in the Request-URI; user agents commonly contact a server in this domain directly in order to deliver a request. However, there is always a possibility that an attacker could impersonate the remote server, and that the user agent's request could be intercepted by some other party. For example, consider a case in which a redirect server at one domain, chicago.com, impersonates a redirect server at another domain, biloxi.com. A user agent sends a request to biloxi.com, but the redirect server at chicago.com answers with a forged response that has appropriate SIP headers for a response from biloxi.com. The forged contact addresses in the redirection response could direct the originating user agent to inappropriate or insecure resources, or simply prevent requests for biloxi.com from succeeding. This family of threats has a vast membership, many of which are critical. As a converse to the registration hijacking threat, consider the case in which a registration sent to biloxi.com is intercepted by chicago.com, which replies to the intercepted registration with a forged 301 (Moved Permanently) response. This response might seem to come from biloxi.com yet designate chicago.com as the appropriate registrar. All future REGISTER requests from the originating user agent would then go to chicago.com. Prevention of this threat requires a means by which user agents can authenticate the servers to whom they send requests. 22.1.3 Tampering with Message Bodies As a matter of course, SIP user agents route requests through trusted proxy servers. Regardless of how that trust is established (authentication of proxies is discussed elsewhere in this section), a user agent may trust a proxy server to route a request, but not to inspect or possibly modify the bodies contained in that request. Consider a UA that is using SIP message bodies to communicate session encryption keys for a media session. Although it trusts the proxy server of the domain it is contacting to deliver signaling properly, it may not be desirable for the administrators of that domain to be capable of decrypting any subsequent media session. Worse yet, if the proxy server were actively malicious, it could modify the session key, either acting as a man-in-the-middle, or perhaps changing the security characteristics requested by the originating user agent. Various Authors [Page 154] Internet Draft SIP January 28, 2002 This family of threats applies not only to session keys, but to most conceivable forms of content carried end-to-end in SIP. These might include MIME bodies that should be rendered to the user, SDP, or encapsulated telephony signals among others. Also note that some headers in SIP are meaningful end-to-end, for example, the Subject. User agents might be protective of these headers as well as bodies (a malicious intermediary changing the Subject header might make an important request appear to be spam, for example). However, since many headers are legitimately inspected or altered by proxy servers as a request is routed, not all headers should be secured end-to-end. For these reasons, the UA might want to secure SIP message bodies, and in some limited cases headers, end-to-end. The security services required for bodies include confidentiality, integrity, and authentication. These end-to-end services should be independent of the means used to secure interactions with intermediaries such as proxy servers. 22.1.4 Tearing Down Sessions Once a dialog has been established by initial messaging, subsequent requests can be sent that modify the state of the dialog and/or session. It is critical that principals in a session can be certain that such requests are not forged by attackers. Consider a case in which a third-party attacker captures some initial messages in a dialog shared by two parties in order to learn the parameters of the session (To, From, and so forth) and then inserts a BYE request into the session. The attacker could opt to forge the request such that it seemed to come from either participant. Once the BYE is received by its target, the session will be torn down prematurely. Similar mid-session threats include the transmission of forged re- INVITEs that alter the session (possibly to reduce session security or redirect media streams as part of a wiretapping attack). The most effective countermeasure to this threat is the authentication of the sender of the BYE - in this instance, the recipient needs only know that the BYE came from the same party with whom the corresponding dialog was established (as opposed to ascertaining the absolute identity of the sender). Also, if the attacker is unable to learn the parameters of the session due to confidentiality, it would not be possible to forge the BYE; however, some intermediaries (like proxy servers) will need to inspect those parameters as the session is established. Various Authors [Page 155] Internet Draft SIP January 28, 2002 22.1.5 Denial of Service and Amplification Denial of service attacks focus on rendering a particular network element unavailable, usually by directing an excessive amount of network traffic at its interfaces. A distributed denial of service attack allows one network user to cause multiple network hosts to flood a target host with a large amount of network traffic. In many architectures SIP proxy servers face the public Internet in order to accept requests from worldwide IP endpoints. SIP creates a number of potential opportunities for distributed denial of service attacks that must be recognized and addressed by the implementers and operators of SIP systems. Attackers can create bogus requests that contain a falsified source IP address and a corresponding Via header field which identify a targeted host as the originator of the request and then send this request to a large number of SIP network elements, thereby using hapless SIP UAs or proxies to generate denial of service traffic aimed at the target. Similarly, attackers might use falsified Route headers in a request that identify the target host and then send such messages to forking proxies that will amplify messaging sent to the target. Record-Route could be used to similar effect when the attacker is certain that the SIP dialog initiated by the request will result in numerous transactions originating in the backwards direction. A number of denial of service attacks open up if REGISTER requests are not properly authenticated and authorized by registrars. Attackers could de-register some or all users in an administrative domain, thereby preventing these users from being invited to new sessions. An attacker could also register a large number of contacts designating the same host for a given address of record in order to use the registrar and any associated proxy servers as amplifiers in a denial of service attack. Attackers might also attempt to deplete available memory and disk resources of a registrar by registering huge numbers of bindings. The use of multicast to transmit SIP requests can greatly increase the potential for denial of service attacks. These problems demonstrate a general need to define architectures that minimize the risks of denial of service, and the need to be mindful in recommendations for security mechanisms of this class of attacks. 22.2 Security Mechanisms Various Authors [Page 156] Internet Draft SIP January 28, 2002 From the threats described above, we gather that the fundamental security services required for the SIP protocol are: preserving the confidentiality and integrity of messaging, preventing replay attacks or message spoofing, providing for the authentication and privacy of the participants in a session, and preventing denial of service attacks. Bodies within SIP messages separately require the security services of: confidentiality, integrity, and authentication. Rather than defining new security mechanisms specific to SIP, SIP reuses wherever possible existing security models derived from the HTTP and SMTP space. Full encryption of messages provides the best means to preserve the confidentiality of signaling - it can also guarantee that messages are not modified by any malicious intermediaries. However, SIP requests and responses cannot be naively encrypted end-to-end in their entirety because, in most network architectures, message fields such as the Request-URI, Route and Via need to be visible to proxies so that SIP requests are routed correctly. Note that proxy servers need to modify some features of messages as well (such as adding Via headers) in order for SIP to function. Proxy servers must therefore be trusted, to some degree, by SIP user agents. To this purpose, low layer security mechanisms for SIP are recommended, which encrypt the entire SIP requests or responses on the wire on a hop-by-hop basis, and which allow endpoints to verify the identity of proxy servers to whom they send requests. SIP entities also have a need to identify one another in a secure fashion. When a SIP endpoint asserts the identity of its user to a peer user agent or to a proxy server, that identity should in some way be verifiable. A cryptographic authentication mechanism is provided in SIP to address this requirement. An independent security mechanism for SIP message bodies supplies an alternative means of end-to-end mutual authentication, as well as providing a limit on the degree to which user agents must trust intermediaries. 22.2.1 Transport and Network Layer Security Transport or network layer security encrypts signaling traffic, guaranteeing message confidentiality and integrity (note however that the originator and recipient of a session may be deducible by observers performing a network traffic analysis). The certificates used to encrypt traffic can also be used to provide a means of authentication in many architectures. Two popular alternatives for providing security at the transport and Various Authors [Page 157] Internet Draft SIP January 28, 2002 network layer are, respectively, TLS [28] and IPSec [29]. IPSec is a set of network-layer protocol tools that can collectively be used as a secure replacement for traditional IP (Internet Protocol). IPSec is most suited to architectures in which a set of SIP hosts (mingled user agents and proxy servers) or bridged administrative domains (possibly using security gateways) have an existing trust relationship with one another, although IPSec can also be used on a per-hop basis. IPSec is generally implemented at an operating-system level within a host, and in many architectures it does not require integration with SIP applications. Any deployment of IPSec for SIP would require an IPSec profile describing the protocols tools that would be required to secure SIP and the modes in which they would operate. No such profile is given in this document. TLS provides transport-layer security over connection-oriented protocols (for the purposes of this document, TCP); "tls" (signifying TLS over TCP) can be specified as the desired transport protocol within a Via header field or a SIP-URI. TLS is most suited to architectures in which a chain of trust joins together a set of hosts. For example, Alice trusts her local proxy server, which in turn trust Bob's local proxy server, which Bob trusts, hence Bob and Alice can communicate securely. TLS must be tightly coupled with a SIP application. Note that transport mechanisms are specified on a hop-by-hop basis in SIP, and that in some deployments TLS might be used for only certain portions of the signaling path. When TLS is used in a SIP application, implementers MUST minimally support the TLS_RSA_WITH_AES_128_CBC_SHA ciphersuite. For purposes of backwards compatibility, proxy servers, redirect servers and registrars SHOULD support TLS_RSA_WITH_3DES_EDE_CBC_SHA. Implementers MAY also support any other ciphersuite. 22.2.2 HTTP Authentication SIP provides a challenge capability, based on HTTP authentication, that relies on the 401 and 407 response codes as well as headers for carrying challenges and credentials. Without significant modification, the reuse of the HTTP Digest authentication scheme in SIP allows for replay protection and one-way authentication. The usage of Digest authentication in SIP is detailed in Section 20. 22.2.3 S/MIME Various Authors [Page 158] Internet Draft SIP January 28, 2002 As is discussed above, encrypting entire SIP messages end-to-end for the purpose of confidentiality is not appropriate because network intermediaries (like proxy servers) need to view certain headers in order to route messages correctly, and if these intermediaries are excluded from security associations then SIP messages will essentially be unroutable. However, S/MIME allows SIP user agents to encrypt MIME bodies within SIP, securing these bodies end-to-end without affecting message headers. S/MIME can provide end-to-end confidentiality and integrity for message bodies, as well as mutual authentication. It is also possible to use S/MIME to provide a form of integrity and confidentiality for SIP headers through SIP message tunneling. The usage of S/MIME in SIP is detailed in Section 21. 22.3 Implementing Security Mechanisms 22.3.1 Requirements for Implementers of SIP Proxy servers, redirect servers, and registrars MUST implement TLS, and MUST support both mutual and one-way authentication. It is strongly RECOMMENDED that user agents be capable initiating TLS; user agents MAY also be capable of acting as a TLS server. Proxy servers, redirect servers, and registrars SHOULD possess a site certificate whose subject corresponds to their hostname. User agents MAY have certificates of their own for mutual authentication with TLS, but no provisions are set forth in this document for their use. User agents MUST support a mechanism for verifying certificates they receive during TLS negotiation. Proxy servers, redirect servers, registrars and user agents MAY also implement IPSec, or other lower-layer security protocols. When a user agent attempts to contact a proxy server, redirect server or registrar, the UAC SHOULD initiate a TLS connection over which it will send SIP messages. In some architectures UACs MAY receive requests over such TLS connections as well. Proxy servers, redirect servers, registrars and user agents MUST implement Digest Authorization. Proxy servers, redirect servers and registrars SHOULD be configured with at least one Digest realm, and at least one "realm" string supported by a given server SHOULD corresponds to the server's hostname or domainname. Proxy servers, redirect servers, registrars and user agents MAY also implement enhancements to Digest or alternate header-level security mechanisms. Various Authors [Page 159] Internet Draft SIP January 28, 2002 User agents SHOULD support S/MIME encryption and signing of SIP message MIME bodies. 22.3.2 Security Solutions The operation of these security mechanisms in concert can follow, to some degree, the existing web and email security models. At a high level, user agents authenticate themselves to servers (proxy servers, redirect servers and registrars) with a Digest username and password; servers authenticate themselves to user agents, and to one another, with a site certificate delivered by TLS. On a peer-to-peer level, user agents ordinarily transitively trust the network to authenticate one another; however, S/MIME can also be used to provide direct authentication when the network does not or if the network itself is not trusted. The following is an illustrative example in which these security mechanisms are used by various user agents and servers to prevent the sorts of threats described in Section 22.1. While implementers and network administrators MAY follow the normative guidelines given in the remainder of this section, these are provided only as example implementations. 22.3.2.1 Registration When a user agent comes on line and registers with its local administrative domain, it SHOULD establish a TLS connection with its registrar (the means by which the user agent determines how to reach its registrar are described in Section 10). The registrar SHOULD offer a certificate to the user agent, and the site identified by the certificate MUST correspond with the domain in which the user agent intends to register; for example, if the user agent intends to register the address of record 'alice@atlanta.com', the site certificate must identify a host within the atlanta.com domain (such as user agent SHOULD verify the certificate and inspect the site identified by the certificate. If the certificate is invalid, revoked, or if it does not identify the appropriate party, the user agent MUST NOT send the REGISTER message and otherwise proceed with the registration. When a valid certificate has been provided by the registrar, the user agent knows that the registrar is not an attacker who might redirect the user agent, steal passwords, or attempt any similar attacks. The user agent then creates a REGISTER request which SHOULD be Various Authors [Page 160] Internet Draft SIP January 28, 2002 addressed to a Request-URI corresponding to the site certificate received from the registrar. When the REGISTER request is sent by the user agent over the existing TLS connection, the registrar SHOULD challenge the request with a 407 (Proxy Authentication Required) response; the "realm" parameter within the Proxy-Authenticate header of the response SHOULD correspond to the domain previously given by the site certificate. When the UAC receives the challenge, it SHOULD either prompt the user for credentials or take an appropriate credential from a keyring corresponding to the "realm" parameter in the challenge. The username of this credential SHOULD correspond with the "userinfo" portion of the URI in the To header of the REGISTER request. Once the Digest credentials have been inserted into an appropriate Proxy-Authorization header, the REGISTER should be resubmitted to the registrar. Since the registrar requires the user agent to authenticate itself, it would be difficult for an attacker to forge REGISTER requests for the user's address of record. Also note that since the REGISTER is sent over a confidential TLS connection, attackers will not be able to intercept the REGISTER to record credentials for any possible replay attack. Once the registration has been accepted by the registrar, the user agent SHOULD leave this TLS connection open provided that the registrar also acts as the proxy server to which requests are sent for users in this administrative domain. The existing TLS connection will be reused to deliver incoming requests to the user agent that has just completed registration. Because the user agent has already authenticated the server on the other side of the TLS connection, all requests that come over this connection are known to have passed through the proxy server - attackers cannot create spoofed requests that appear to have been sent through that proxy server. 22.3.2.2 Requests and Transitive Trust Now let's say that Alice's user agent would like to initiate a session with a user in a remote administrative domain, namely 'bob@biloxi.com'. We'll also say that the local administrative domain ('atlanta.com') has a local outbound proxy. The proxy server that handles inbound requests for an administrative domain MAY also act as a local outbound proxy; for simplicity's sake we'll assume this to be the case for 'atlanta.com' (otherwise the Various Authors [Page 161] Internet Draft SIP January 28, 2002 user agent would initiate a new TLS connection to a separate server at this point). Assuming that the client has completed the registration process described in the preceding section, it SHOULD reuse the TLS connection to the local proxy server when it wishes to send an INVITE request to another user. The user agent SHOULD reuse cached credentials in the INVITE to avoid prompting the user unnecessarily. When the local outbound proxy server has validated the credentials presented by the user agent in the INVITE, it SHOULD inspect the Request-URI to determine how the message should be routed (see [8]). If the "domainname" portion of the Request-URI had corresponded to the local domain ('atlanta.com'), rather the "biloxi.com", then the proxy server would have consulted its location service to determine how best to reach the requested user. Had 'alice@atlanta.com' been attempting to contact, say, the TLS connection Alex had established with the register when he registered. Since Alex would receive this request over his authenticated channel, he would be assured that Alice's request had been authorized by the proxy server of the local administrative domain. However, in this instance the Request-URI designates a remote domain. The local outbound proxy server at 'atlanta.com' SHOULD therefore establish a TLS connection with the remote proxy server at servers that possess site certificates, mutual TLS authentication SHOULD occur. Each side of the connection SHOULD verify and inspect the certificate of the other, noting the domain name that appears in the certificate for comparison with the headers of SIP messages. The 'atlanta.com' proxy server, for example, SHOULD verify at this stage that the certificate received from the remote side corresponds with the 'biloxi.com' domain. Once it has done so, and TLS negotiation has completed, resulting in a secure channel between the two proxies, the 'atlanta.com' proxy can forward the INVITE request to The proxy server at 'biloxi.com' SHOULD in turn inspect the certificate of the proxy server at 'atlanta.com' and compare the domain asserted by the certificate with the "domainname" portion of the From header in the INVITE request. The biloxi proxy can thereby ascertain whether or not it should consider Alice to be transitively authenticated. The biloxi proxy MAY have a strict security policy that requires it to reject requests that do not match the administrative domain from which they have been proxied, or perhaps even more strictly, requests that originate from administrative domains that do not have some policy agreement with biloxi. Various Authors [Page 162] Internet Draft SIP January 28, 2002 Such security policies could be instituted to prevent the SIP equivalent of SMTP 'open relays' which are frequently exploited to generate spam. Once the INVITE has been approved by the biloxi proxy, the proxy server SHOULD identify the existing TLS channel, if any, associated with the user targeted by this request (in this case 'bob@biloxi.com'). The INVITE should be proxied through this channel to Bob; since the request is received over a TLS connection which had previously been authenticated as the biloxi proxy, Bob transitively trusts the identity asserted in the From header. Before they forward the request, both proxy servers SHOULD add Record-Route headers to the request so that all future requests in this dialog will pass through the proxy servers. The proxy servers can thereby continue to provide transitive authentication, confidentiality, replay protection, and so forth for lifetime of this dialog. If the proxy servers do not add themselves to the Record- Route, future messages will pass directly end-to-end between Alice and Bob without any security services (unless the two parties agree on some independent end-to-end security). An attacker preying on this architecture would, for example, be unable to forge a BYE request and insert it into the signaling stream between Bob and Alice because the attacker has no way of ascertaining the parameters of the session because of the use of confidentiality, and moreover because the integrity mechanism transitively protects the traffic all the way from Alice to Bob. 22.3.2.3 Peer to Peer Requests Alternatively, consider a user agent asserting the identity to send an INVITE to 'bob@biloxi.com', her user agent SHOULD initiate a TLS connection with the biloxi proxy directly (using the mechanism described in [8] to determine how to best to reach the given Request-URI). When her user agent receives a certificate from the biloxi proxy, it SHOULD be verified normally before she passes her INVITE across the TLS connection. However, proxy; but she does have a CMS detached signature over a "message/sip" body in the INVITE. It is unlikely in this instance that Carol would have any credentials in the 'biloxi.com' realm, since she has no formal association with biloxi.com. The biloxi proxy MAY also have a strict policy that precludes it from even bothering to challenge requests that do not have 'biloxi.com' in the "domainname" portion of the From header - it treats these users as unauthenticated. Various Authors [Page 163] Internet Draft SIP January 28, 2002 The biloxi proxy has a policy for Bob that all non-authenticated requests should be redirected to the appropriate contact address registered against 'bob@biloxi.com', namely . Carol receives the redirection response over the TLS connection she established with the biloxi proxy, so she trusts the veracity of the contact address. Carol SHOULD then established a TCP connection with the designated address and send a new INVITE with a Request-URI containing the received contact address (recomputing the signature in the body as the request is readied). Bob receives this INVITE on an insecure interface, but his user agent inspects and in this instance recognizes the From header of the request and subsequently matches a locally cached certificate with the one presented in the signature of the body of the INVITE. He replies in similar fashion, authenticating himself to Carol, and a secure dialog begins. Sometimes firewalls or NATs in an administrative domain could preclude the establishment of a direct TCP connection to a user agent. In these cases, proxy servers could also potentially relay requests to user agents in a way that has no trust implications (for example, forgoing an existing TLS connection and forwarding the request over cleartext TCP) as local policy dictates. 22.3.2.4 DoS Protection In order to minimize the risk of a denial of service attack against architectures using these security solutions, implementers should take note of the following guidelines. When the host on which a SIP proxy server is operating is routable from the public Internet, it SHOULD be deployed in an administrative domain with secure routing policies (blocking source-routed traffic, preferably filtering ping traffic). Both TLS and IPSec can also make use of bastion hosts at the edges of administrative domains that participate in the security associations to aggregate secure tunnels and sockets. These bastion hosts can also take the brunt of denial of service attacks, ensuring that SIP hosts within the administrative domain are not encumbered with superfluous messaging. No matter what security solutions are deployed, floods of messages directed at proxy servers can lock up proxy server resources and prevent desirable traffic from reaching its destination. There is a computational expense associated with processing a SIP transaction at a proxy server, and that expense is greater for stateful proxy servers than it is for stateless proxy servers. Therefore stateful Various Authors [Page 164] Internet Draft SIP January 28, 2002 proxies are more susceptible to flooding than stateless proxy servers. User agents and proxy servers SHOULD challenge questionable requests with only a single 401 (Unauthorized) or 407 (Proxy Authentication Required), forgoing the normal response retransmission algorithm, and behaving statelessly towards unauthenticated requests. Retransmitting the 401 (Unauthorized) or 407 (Proxy Authentication Required) status response amplifies the problem of an attacker using a falsified header (such as Via) to direct traffic to a third party. With either TCP or UDP, a denial of service attack exists by a rogue proxy sending 6xx responses. Although a client SHOULD choose to ignore such responses if it requested authentication, a proxy cannot do so. It is obliged to forward the 6xx response back to the client. The client can then ignore the response, but if it repeats the request it will probably reach the same rogue proxy again, and the process will repeat. 22.4 Limitations Although these security mechanisms, when applied in a judicious manner, can thwart many threats, there are limitations in the scope of the mechanisms that must be understood by implementers and network operators. 22.4.1 HTTP Digest One of the primary limitations of using HTTP Digest in SIP is that the integrity mechanisms in Digest do not work very well for SIP. Specifically, they offer protection of the Request-URI and the method of a message, but not for any of the headers that user agents would most likely wish to secure. The existing replay protection mechanisms described in RFC 2617 also have some limitations for SIP. The next-nonce mechanism, for example, does not support pipelined requests. The nonce-count mechanism should be used for replay protection. Another limitation of HTTP Digest is the scope of realms. Digest is valuable when a user wants to authenticate themselves to a resource with which they have a pre-existing association, like a service provider of which the user is a customer. Consider that by contrast, the scope of TLS is global, since certificates are globally verifiable regardless of any pre-existing association between the user agent and the server. Various Authors [Page 165] Internet Draft SIP January 28, 2002 Future enhancements to HTTP Digest could conceivably resolve some or all of these limitations. 22.4.2 S/MIME The largest outstanding defect with the S/MIME mechanism is the lack of prevalent public key infrastructure for end users. If self-signed certificates (or certificates that cannot be verified by one of the participants in a dialog) are used, the SIP-based key exchange mechanism described in Section 21.2 is susceptible to a man-in-the- middle attack with which an attacker can potentially inspect and modify S/MIME bodies. The attacker needs to intercept the first exchange of keys between the two parties in a dialog, remove the existing CMS detached signatures from the request and response, and insert a different CMS detached signature containing a certificate supplied by the attacker (but which seems to be a certificate for the proper address of record). Each party will think they have exchanged keys with the other, when in fact each has the public key of the attacker. It is important to note that the attacker can only leverage this vulnerability on the first exchange of keys between two parties - on subsequent occasions, the alteration of the key would be noticeable to user agents. It would also be difficult for the attacker to remain in the path of all future dialogs between the two parties over time (as potentially days, weeks, or years pass). SSH is susceptible to the same man-in-the-middle attack on the first exchange of keys; however, it is widely acknowledged that while SSH is not perfect, it does improve the security of connections. The use of key fingerprints could provide some assistance to SIP, just as it does for SSH. For example, if two parties use SIP to establish a voice communications session, each could read off the fingerprint of the key they received from the other, which could be compared against the original; it would certainly be more difficult for the man-in- the-middle to emulate the voices of the participants than their signaling. The S/MIME mechanism allows user agents to send encrypted requests without preamble if they possess a certificate for the destination address of record on their keyring. However, it is also possible that a device which does not hold certificates, or at least not that particular certificate, will be currently registered as the sole contact address for that address of record, and it will therefore be unable to properly process the encrypted request, which could lead to some avoidable error signaling. This is especially likely when an encrypted request is forked. Various Authors [Page 166] Internet Draft SIP January 28, 2002 The keys associated with S/MIME are most useful when associated with a particular user (an address of record) rather than a device (a user agent). When users move between devices, it may be difficult to transport private keys securely between user agents; how such keys might be acquired by a device is outside the scope of this document. Another, more prosaic difficulty with the S/MIME mechanism is that it can result in very large messages, especially when the SIP tunneling mechanism described in Section 21.4 is used. For that reason, it is RECOMMENDED that TCP should be used as a transport protocol when S/MIME tunneling is employed. 22.4.3 TLS The most commonly voiced concern about TLS is that it cannot run over UDP; TLS requires a connection-oriented underlying transport protocol, which for the purposes of this document means TCP. Even running TCP, regardless of any additional overhead incurred by TLS, is argued to be too intensive for some embedded devices. It may also be arduous for a local outbound proxy server and/or registrar to maintain many simultaneous long-lived TLS connections with numerous user agents might. This introduces some valid scalability concerns, especially for intensive ciphersuites. Maintaining redundancy of long-lived TLS connections, especially when a user agent is solely responsible for their establishment, could also be cumbersome. TLS only allows SIP entities to authenticate servers to which they are adjacent; TLS offers strictly hop-by-hop security. Neither TLS, nor any other mechanism specified in this document, allows clients to authenticate proxy servers to whom they cannot form a direct TCP connection. 22.5 Privacy SIP messages frequently contain sensitive information about their senders - not just what they have to say, but with whom they communicate, when they communicate and for how long, and from where they participate in sessions. Many applications and their users require that this sort of private information be hidden from any parties that do not need to know it. Note that there are also less direct ways in which private information can be divulged. If a user or service chooses to be reachable at an address that is guessable from the person's name and organizational affiliation (which describes most addresses of record), the traditional method of ensuring privacy by having an Various Authors [Page 167] Internet Draft SIP January 28, 2002 unlisted "phone number" is compromised. A user location service can infringe on the privacy of the recipient of a session invitation by divulging their specific whereabouts to the caller; an implementation consequently SHOULD be able to restrict, on a per-user basis, what kind of location and availability information is given out to certain classes of callers. 23 Common Message Components There are certain components of SIP messages that appear in various places within SIP messages (and sometimes, outside of them), which merit separate discussion. 23.1 SIP Uniform Resource Indicators A SIP URI identifies a communications resource. Like all URIs, SIP URIs may be placed in web pages, email messages or printed literature. They contain sufficient information to initiate and maintain a communication session with the resource. Examples of communications resources include o a user of an online service; o an appearance on a multiline phone; o a mailbox on a messaging system o a PSTN number at a gateway service; o a group (such as "sales" or "helpdesk") in an organization. 23.1.1 SIP URI Components The "sip:" scheme follows the guidelines in RFC 2396 [9]. It uses a form similar to the mailto URL, allowing the specification of SIP request-header fields and the SIP message-body. This makes it possible to specify the subject, media type, or urgency of sessions initiated by using a URI on a web page or in an email message. The formal syntax for a SIP URI is presented in Section 27. Its general form is sip:user:password@host:port;url-parameters?headers have the following meaning. user: The identifier of a particular resource at the host being addressed. Note that "host" as used here may, and frequently does, refer to a domain. The "userpart" of a URI consists of this user field, the password field and the @ Various Authors [Page 168] Internet Draft SIP January 28, 2002 sign following them. The userpart of a URI is optional and MAY be absent when the destination host does not have a notion of users or when the host itself is the resource being identified. If the @ sign is present in a SIP URI, the user field MUST NOT be empty. If the host being addressed is capable of processing telephone numbers, an Internet telephony gateway for instance, a telephone-subscriber field defined in RFC 2806 [13] MAY be used to populate the user field. There are special escaping rules for encoding telephone-subscriber fields in SIP URIs described in Section 23.1.2. password: A password associated with the user. While the SIP URI syntax allows this field to be present, its use is NOT RECOMMENDED, because the passing of authentication information in clear text (such as URIs) has proven to be a security risk in almost every case where it has been used. For instance, transporting a PIN number in this field exposes the PIN. Note that the password field is just an extension of user portion. Implementations not wishing to give special significance to the password portion of the field MAY simply treat "user:password" as a single string. host: The entity hosting the SIP resource. The host part contains either a fully-qualified domain name or numeric IPv4 or IPv6 address. Using the fully-qualified domain name form is RECOMMENDED whenever possible. port: The port number where the request is to be sent. URI parameters: Parameters affecting a request constructed from the URI. URI parameters are added after the hostport component and are separated by semi-colons. URI parameters take the form: parameter-name "=" parameter-value Even though an arbitrary number of URI parameters may be included in a URI, any given parameter-name MUST NOT appear more than once. This extensible mechanism includes the transport, maddr, ttl, user, and method parameters. The transport parameter determines the transport mechanism to be used for sending SIP messages, as specified in [8]. SIP can use any network transport protocol. Parameter Various Authors [Page 169] Internet Draft SIP January 28, 2002 names are defined for UDP [30], TCP [31], TLS [28] (note that this is specifically TLS over TCP), and SCTP [32]. The maddr parameter indicates the server address to be contacted for this user, overriding any address derived from the host field. When an maddr parameter is present, the port and transport components of the URI apply to the address indicated in the maddr parameter value. [8] describes the proper interpretation of the transport, maddr and hostport in order to obtain the destination address, port and transport for sending a request. The maddr field can be used as a simple form of loose source routing. It allows a URI to specify a specific proxy that must be traversed en-route to the destination. This capability is useful for a roaming user that is forced to use an outbound proxy, but wishes to force requests through their home proxy. Alternatively, preloaded Route values can be used to provide this capability (see item 8.1.1.1 in section 8.1.1). The ttl parameter determines the time-to-live value of the UDP multicast packet and MUST only be used if maddr is a multicast address and the transport protocol is UDP. For example, to specify to call alice@atlanta.com using multicast to 239.255.255.1 with a ttl of 15, the following URI would be used: sip:alice@atlanta.com;maddr=239.255.255.1;ttl=15 The set of valid telephone-subscriber strings is a subset of valid user strings. The user URI parameter exists to distinguish telephone numbers from user names that happen to look like telephone numbers. If the user string contains a telephone number formatted as a telephone- subscriber, the user parameter value "phone" SHOULD be present. Even without this parameter, recipients of SIP URIs MAY interpret the pre-@ part as a telephone number if local restrictions on the name space for user name allow it. The method of the SIP request constructed from the URI can be specified with the method parameter. Various Authors [Page 170] Internet Draft SIP January 28, 2002 Since the url-parameter mechanism is extensible, SIP elements MUST silently ignore any url-parameters that they do not understand. Headers: Headers to be included in a request constructed from the URI. Headers fields in the SIP request can be specified with the "?" mechanism within a SIP URI. The header names and values are encoded in ampersand separated hname = hvalue pairs. The special hname "body" indicates that the associated hvalue is the message-body of the SIP request. Table 1 summarizes the use of SIP URI components based on the context in which the URI appears. The external column describes URIs appearing anywhere outside of a SIP message, for instance on a web page or business card. Entries marked "m" are mandatory, those marked "o" are optional, and those marked "-" are not allowed. Elements processing URIs SHOULD ignore any disallowed components if they are present. The second column indicates the default value of an optional element if it is not present. "--" indicates that the element is either not optional, or has no default value. SIP URIs in Contact header fields have different restrictions depending on the context in which the header field appears. One set applies to messages that establish and maintain dialogs (INVITE and its 200 (OK) response). The other applies to registration and redirection messages (REGISTER, its 200 (OK) response, and 3xx class responses to any method). dialog reg./redir. Contact/ default Req.-URI To From Contact R-R/Route external user -- o o o o o o password -- o o o o o o host -- m m m m m m port 5060 o - - o o o user-param ip o o o o o o method INVITE - - - - - o maddr-param -- o - - o o o ttl-param 1 o - - o - o transp.-param udp o - - o o o other-param -- o o o o o o headers -- - - - o - o Table 1: Use and default values of URI components for SIP headers, Request-URI and references Various Authors [Page 171] Internet Draft SIP January 28, 2002 23.1.2 Character Escaping Requirements SIP follows the requirements and guidelines of RFC 2396 [9] when defining the set of characters that must be escaped in a SIP URI, and uses its ""%" HEX HEX" mechanism for escaping. From RFC 2396: The set of characters actually reserved within any given URI component is defined by that component. In general, a character is reserved if the semantics of the URI changes if the character is replaced with its escaped US-ASCII encoding. [9]. Excluded US-ASCII characters [9], such as space and control characters and characters used as URI delimiters, also MUST be escaped. URIs MUST NOT contain unescaped space and control characters. For each component, the set of valid BNF expansions defines exactly which characters may appear unescaped. All other characters MUST be escaped. For example, "@" is not in the set of characters in the user component, so the user "j@s0n" must have at least the @ sign encoded, as in "j%40s0n". Expanding the hname and hvalue tokens in Section 27 show that all URI reserved characters in header names and values MUST be escaped. The telephone-subscriber subset of the user component has special escaping considerations. The set of characters not reserved in the RFC 2806 [13] description of telephone-subscriber contains a number of characters in various syntax elements that need to be escaped when used in SIP URIs. Any characters occurring in a telephone-subscriber that do not appear in an expansion of the BNF for the user rule MUST be escaped. Note that character escaping is not allowed in the host component of a SIP URI (the % character is not valid in its expansion). This is likely to change in the future as requirements for Internationalized Domain Names are finalized. Current implementations MUST NOT attempt to improve robustness by treating received escaped characters in the host component as literally equivalent to their unescaped counterpart. The behavior required to meet the requirements of IDN may be significantly different. 23.1.3 Example SIP URIs sip:alice@atlanta.com Various Authors [Page 172] Internet Draft SIP January 28, 2002 sip:alice:secretword@atlanta.com;transport=tcp sip:alice@atlanta.com?subject=project sip:+1-212-555-1212:1234@gateway.com;user=phone sip:1212@gateway.com sip:alice@192.0.2.4 sip:atlanta.com;method=REGISTER?to=alice sip:alice;day=tuesday@atlanta.com The last example URI above has a user field value of "alice;day=tuesday". The escaping rules defined above allow a semicolon to appear unescaped in this field. Note, however, that for the purposes of this protocol, the field is opaque. The apparent structure in that value is only useful to the entity responsible for the resource. 23.1.4 SIP URI Comparison SIP URIs are compared for equality according to the following rules: o Comparison of the userpart of sip URIs is case-sensitive. This includes userparts containing passwords or formatted as telephone-subscribers. Comparison of all other components of the URI is case-insensitive unless explicitly defined otherwise. o The ordering of parameters and headers is not significant in comparing SIP URIs. o Characters other than those in the "reserved" and "unsafe" sets (see RFC 2396 [9]) are equivalent to their ""%" HEX HEX" encoding. o An IP address that is the result of a DNS lookup of a host name does not match that host name. o For two URIs to be equal, the user, password, host, and port components must match. A URI omitting the optional port component will match a URI explicitly declaring port 5060. A URI omitting the user component will not match a URI that includes one. A URI omitting the password component will not match a URI that includes one. o URI uri-parameter components are compared as follows - Any uri-parameter appearing in both URIs must match. Various Authors [Page 173] Internet Draft SIP January 28, 2002 - A user, transport, ttl, or method url-parameter appearing in only one URI must contain its default value or the URIs do not match. A URI that includes an maddr parameter will not match a URI that contains no maddr parameter. - All other url-parameters appearing in only one URI are ignored when comparing the URIs. o URI header components are never ignored. Any present header component MUST be present in both URIs and match for the URIs to match. The matching rules are defined for each header in Section sec:header-fields. The URIs within each of the following sets are equivalent: sip: sip:alice@AtLanTa.CoM;Transport=udp sip:carol@chicago.com sip:carol@chicago.com;newparam=5 sip:carol@chicago.com;security=on sip:biloxi.com;transport=tcp;method=REGISTER?to=sip:bob sip:biloxi.com;method=REGISTER;transport=tcp?to=sip:bob sip:alice@atlanta.com?subject=project sip:alice@atlanta.com?priority=urgent&subject=project The URIs within each of the following sets are not equivalent: SIP:ALICE@AtLanTa.CoM;Transport=udp (different usernames) sip:alice@AtLanTa.CoM;Transport=UDP Various Authors [Page 174] Internet Draft SIP January 28, 2002 sip:bob@biloxi.com (different port and transport) sip:bob@biloxi.com:6000;transport=tcp sip:carol@chicago.com (different header component) sip:carol@chicago.com?Subject=next sip:bob@phone21.boxesbybob.com (even though that's what sip:bob@192.0.2.4 phone21.boxesbybob.com resolves to) Note that equality is not transitive: o sip:carol@chicago.com and sip:carol@chicago.com;security=on are equivalent o sip:carol@chicago.com and sip:carol@chicago.com;security=off are equivalent o sip:carol@chicago.com;security=on and sip:carol@chicago.com;security=off are not equivalent Comparing URIs is a major part of comparing several SIP headers (see Section 24). 23.1.5 Forming Requests from a SIP URI An implementation must take care when forming requests directly from a URI. URIs from business cards, web pages, and even from sources inside the protocol such as registered contacts may contain inappropriate header fields or body parts. An implementation MUST include any provided transport, maddr, ttl, or user parameter in the Request-URI of the formed request. If the URI contains a method parameter, its value MUST be used as the method of the request. The method parameter MUST NOT be placed in the Request- URI. Unknown URI parameters MUST be placed in the message's Request- URI. An implementation SHOULD treat the presence of any headers or body parts in the URI as a request to include them in the message, and choose to honor the request on an per-component basis. Various Authors [Page 175] Internet Draft SIP January 28, 2002 An implementation SHOULD NOT honor these obviously dangerous header fields: From, Call-ID, CSeq, Via, and Record-Route. An implementation SHOULD take special care in honoring any requested Route header field values in order to not be used as an unwitting agent in malicious attacks. An implementation SHOULD NOT honor requests to include headers that may cause it to falsely advertise its location or capabilities. These include: Accept, Accept-Encoding, Accept-Language, Allow, Contact (in its dialog usage), Organization, Supported, and User-Agent. An implementation SHOULD verify the accuracy of any requested descriptive headers, including: Content-Disposition, Content- Encoding, Content-Language, Content-Length, Content-Type, Date, Mime-Version, and Timestamp. If the request formed from constructing a message from a given URI is not a valid SIP request, the URI is invalid. An implementation MUST NOT proceed with transmitting the request. It should instead pursue the course of action due an invalid URI in the context it occurs. The constructed request can be invalid in many ways. These include, but are not limited to, syntax error in header fields, invalid combinations of URI parameters, or an incorrect description of the message body. Sending a request formed from a given URI may require capabilities unavailable to the implementation. The URI might indicate use of an unimplemented transport or extension for example. An implementation SHOULD refuse to send these requests rather than modifying them to match their capabilities. An implementation MUST NOT send a request requiring an extension that it does not support. For example, such a request can be formed through the presence of a headerRequire header parameter or a method URI parameter with an unknown or explicitly unsupported value. 23.1.6 Relating SIP URIs and tel URLs When a tel URL [13] is converted to a SIP URI, the entire telephone- subscriber portion of the tel URL, including any paramters,is placed into the userpart of the SIP URI. Thus, tel:+358-555-1234567;postd=pp22 becomes Various Authors [Page 176] Internet Draft SIP January 28, 2002 sip:+358-555-1234567;postd=pp22@foo.com not sip:+358-555-1234567@foo.com;postd=pp22 In general, equivalent "tel" URLs converted to SIP URIs in this fashion may not produce equivalent SIP URIs. The userpart of SIP URIs is compared as a case-sensitive string. Variance in case-insensitive portions of tel URLs and reordering of tel URL parameters does not affect tel URL equivalence, but does affect the equivalence of SIP URIs formed from them. For example, tel:+358-555-1234567;postd=pp22 tel:+358-555-1234567;POSTD=PP22 are equivalent, while sip:+358-555-1234567;postd=pp22@foo.com sip:+358-555-1234567;POSTD=PP22@foo.com are not. Likewise, tel:+358-555-1234567;postd=pp22;isub=1411 tel:+358-555-1234567;isub=1411;postd=pp22 are equivalent, while sip:+358-555-1234567;postd=pp22;isub=1411@foo.com sip:+358-555-1234567;isub=1411;postd=pp22@foo.com are not. To mitigatate this problem, elements constructing telephone- subscriber fields to place in the userpart of a SIP URI SHOULD fold any case-insensitive portion of telephone-subscriber to lower case, and order the telephone-subscriber parameters lexically by parameter Various Authors [Page 177] Internet Draft SIP January 28, 2002 name. (All components of a tel URL except for future-extension parameters are defined to be compared case-insensitive.) Following this suggestion, both tel:+358-555-1234567;postd=pp22 tel:+358-555-1234567;POSTD=PP22 become sip:+358-555-1234567;postd=pp22@foo.com and both tel:+358-555-1234567;postd=pp22;isub=1411 tel:+358-555-1234567;isub=1411;postd=pp22 become sip:+358-555-1234567;isub=1411;postd=pp22 23.2 Option Tags Option tags are unique identifiers used to designate new options (extensions) in SIP. These tags are used in Require (Section 24.33), Proxy-Require (Section 24.29, Supported (Section 24.39) and Unsupported (Section 24.42) header fields. Note that these options appear as parameters in those headers in an option-tag = token form (see Section 27 for the definition of token). The creator of a new SIP option MUST either prefix the option with their reverse domain name or register the new option with the Internet Assigned Numbers Authority (IANA) (See Section 28). An example of a reverse-domain-name option is "com.foo.mynewfeature", whose inventor can be reached at "foo.com". For these features, individual organizations are responsible for ensuring that option names do not collide within the same domain. The host name part of the option MUST use lower-case; the option name is case-insensitive. Options registered with IANA do not contain periods and are globally unique. IANA option tags are case-insensitive. Various Authors [Page 178] Internet Draft SIP January 28, 2002 23.3 Tags The "tag" parameter is used in the To and From fields of SIP messages. It serves as a general mechanism to identify a particular instance of a user agent for a particular SIP URI. As proxies can fork requests, the same request can reach multiple instances of a user (mobile and home phones, for example). Since each can respond, there needs to be a means for the originator of a session to distinguish the responses. Tag fields in the To and From disambiguate these multiple instances of the same user. This situation also arises with multicast requests. When a tag is generated by a UA for insertion into a request or response, it MUST be globally unique and cryptographically random with at least 32 bits of randomness. A property of this selection requirement is that a UA will place a different tag into the From header of an INVITE as it would place into the To header of the response to the same INVITE. This is needed in order for a UA to invite itself to a session, a common case for "hairpinning" of calls in PSTN gateways. Similarly, two INVITEs for different calls will have different From tags. Besides the requirement for global uniqueness, the algorithm for generating a tag is implementation specific. Tags are helpful in fault tolerant systems, where a dialog is to be recovered on an alternate server after a failure. A UAS can select the tag in such a way that a backup can recognize a request as part of a dialog on the failed server, and therefore determine that it should attempt to recover the dialog and any other state associated with it. 24 Header Fields The general syntax for header fields is covered in Section 7.3. This section lists the full set of header fields along with notes on syntax, meaning, and usage. Throughout this section, we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1 specification RFC 2616 [12]. Examples of each header field are given. Information about header fields in relation to methods and proxy processing is summarized in Tables 2 and 3. The "where" column describes the request and response types in which the header field can be used. Values in this column are: R: header fields may only appear in requests; Various Authors [Page 179] Internet Draft SIP January 28, 2002 r: header field may only appear in responses; 2xx, 4xx, etc.: A numerical value or range indicates response codes with which the header field can be used; c: header field is copied from the request to the response. An empty entry in the "where" column indicates that the header may be present in all requests and responses. The "proxy" column describes the operations a proxy may perform on a header: c: A proxy can add (concatenate) comma-separated elements to the header. m: A proxy can modify the header. a: A proxy can add the header if not present. r: A proxy must be be able to read the header and thus this header cannot be encrypted. The next six columns relate to the presence of a header field in a method: o: The header field is optional. m: The header field is mandatory. m*: The header field SHOULD be sent, but servers need to be prepared to receive messages without that header field. t: The header field SHOULD be sent, but servers need to be prepared to receive messages without that header field. If TCP is used as transport, then the header field MUST be sent. *: The header field is required if the message body is not empty. See sections 24.14, 24.15 and 7.4 for details. -: The header field is ignored. c: Conditional; the header field is either mandatory or optional, depending on the presence of a route set or the response code. "Optional" means that a UA MAY include the header field in a request Various Authors [Page 180] Internet Draft SIP January 28, 2002 or response, and a UA MAY ignore the header field if present in the request or response (The exception to this rule is the Require header field discussed in 24.33). A "mandatory" header field MUST be present in a request, and MUST be understood by the UAS receiving the request. A mandatory response header field MUST be present in the response, and the header field MUST be understood by the UAC processing the response. "Not applicable" means that the header field MUST NOT be present in a request. If one is placed in a request by mistake, it MUST be ignored by the UAS receiving the request. Similarly, a header field labeled "not applicable" for a response means that the UAS MUST NOT place the header in the response, and the UAC MUST ignore the header in the response. A UA SHOULD ignore extension header parameters that are not understood. A compact form of some common header fields is also defined for use when overall message size is an issue. The Contact, From, and To header fields contain a URI. If the URI contains a comma, question mark or semicolon, the URI MUST be enclosed in angle brackets (< and >). Any URI parameters are contained within these brackets. If the URI is not enclosed in angle brackets, any semicolon-delimited parameters are header-parameters, not URI parameters. 24.1 Accept The Accept header follows the syntax defined in [H14.1]. The semantics are also identical, with the exception that if no Accept header is present, the server SHOULD assume a default value of application/sdp An empty Accept header means that no formats are acceptable. Example: Accept: application/sdp;level=1, application/x-private, text/html 24.2 Accept-Encoding The Accept-Encoding header field is similar to Accept, but restricts the content-codings [H3.5] that are acceptable in the response. See [H14.3]. The syntax of this header is defined in [H14.3]. The semantics in SIP are identical to those defined in [H14.3]. Various Authors [Page 181] Internet Draft SIP January 28, 2002 Header field where proxy ACK BYE CAN INV OPT REG PRA _______________________________________________________________ Accept R - o - m* m* o o Accept 2xx - - - m* m* o - Accept 415 - o - o o o o Accept-Encoding R - o - m* o o o Accept-Encoding 2xx - - - m* m* o - Accept-Encoding 415 - o - o o o o Accept-Language R - o - m* o o o Accept-Language 2xx - - - m* m* o - Accept-Language 415 - o - o o o o Alert-Info R am - - - o - - - Alert-Info 180 am - - - o - - - Allow R o o o o o o o Allow 2xx - o o m* m* o o Allow r - o o o o o o Allow 405 - m m m m m m Authentication-Info 2xx - o - o o o o Authorization R o o o o o o o Call-ID c r m m m m m m m Call-Info am - - - o o o - Contact R o - - m o o - Contact 1xx - - - o o - - Contact 2xx - - - m o o - Contact 3xx - o - o o o o Contact 485 - o - o o o o Content-Disposition o o - o o o o Content-Encoding o o - o o o o Content-Language o o - o o o o Content-Length r t t t t t t t Content-Type * * - * * * * CSeq c r m m m m m m m Date a o o o o o o o Error-Info 300-699 - o o o o o o Expires - - - o - o - From c r m m m m m m m In-Reply-To R - - - o - - - Max-Forwards R amr m m m m m m m Min-Expires 423 - - - - - m - MIME-Version o o o o o o o Organization am - - - o o o - Table 2: Summary of header fields, A--O Various Authors [Page 182] Internet Draft SIP January 28, 2002 Header field where proxy ACK BYE CAN INV OPT REG PRA _______________________________________________________________________ Priority R a - - - o - - - Proxy-Authenticate 407 - m m m m m m Proxy-Authorization R r o o o o o o o Proxy-Require R r - o - o o o o RAck R - - - - - - m Record-Route R amr o o o o o - o Record-Route 2xx,401,484 - o o o o - o Reply-To - - - o - - - Require acr - o - o o o o Retry-After 404,413,480,486 - o o o o o o 500,503 - o o o o o o 600,603 - o o o o o o Route R r c c c c c - c RSeq 1xx - o - o o o - Server r - o o o o o o Subject R - - - o - - - Supported R - o o o o o o Supported 2xx - o o o m* o o Timestamp o o o o o o o To c(1) r m m m m m m m Unsupported 420 - o o o o o o User-Agent o o o o o o o Via c acmr m m m m m m m Warning r - o o o o o o WWW-Authenticate 401 - m m m m m m Table 3: Summary of header fields, P--Z; (1): copied with possible addition of tag An empty Accept-Encoding header field is permissible, even though the syntax in [H14.3] does not provide for it. It is equivalent to Accept-Encoding: identity, that is, only the identity encoding, meaning no encoding, is permissible. If no Accept-Encoding header is present, the server SHOULD assume a default value of identity. This differs slightly from the HTTP definition, which indicates that when not present, any encoding can be used, but the identity encoding is preferred. Example: Accept-Encoding: gzip Various Authors [Page 183] Internet Draft SIP January 28, 2002 24.3 Accept-Language The Accept-Language header is used in requests to indicate the preferred languages for reason phrases, session descriptions, or status responses carried as message bodies in the response. If no Accept-Language header is present, the server SHOULD assume all languages are acceptable to the client. The Accept-Language header follows the syntax defined in [H14.4]. The rules for ordering the languages based on the "q" parameter apply to SIP as well. Example: Accept-Language: da, en-gb;q=0.8, en;q=0.7 24.4 Alert-Info When present in an INVITE request, the Alert-Info header field specifies an alternative ring tone to the UAS. When present in a 180 (Ringing) response, the Alert-Info header field specifies an alternative ringback tone to the UAC. A typical usage is for a proxy to insert this header to provide a distinctive ring feature. The Alert-Info header can introduce security risks. These risks and the ways to handle them are discussed in Section 24.9, which discusses the Call-Info header since the risks are identical. In addition, a user SHOULD be able to disable this feature selectively. This helps prevent disruptions that could result from the use of this header by untrusted elements. Example: Alert-Info: 24.5 Allow The Allow header field lists the set of methods supported by the UA generating the message. All methods, including ACK and CANCEL, understood by the UA MUST be Various Authors [Page 184] Internet Draft SIP January 28, 2002 included in the list of methods in the Allow header, when present. The absence of an Allow header MUST NOT be interpreted to mean that the UA sending the message supports no methods. Rather, it implies that the UA is not providing any information on what methods it supports. Supplying an Allow header in responses to methods other than OPTIONS reduces the number of messages needed. Example: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE 24.6 Authentication-Info The Authentication-Info header provides for mutual authentication with HTTP Digest. A UAS MAY include this header in a 2xx response to a request that was successfully authenticated using digest based on the Authorization header. Syntax and semantics follow those specified in RFC 2617 [23]. Example: Authentication-Info: nextnonce="47364c23432d2e131a5fb210812c" 24.7 Authorization The Authorization header field contains authentication credentials of a UA. Section 20.2 overviews the use of the Authorization header field, and Section 20.4 describes the syntax and semantics when used with HTTP authentication. This header field, along with Proxy- Authorization, breaks the general rules about multiple header fields. Although not a comma-separated list, this header field may be present multiple times, and MUST NOT be combined into a single header using the usual rules described in Section 7.3. In the example below, there are no quotes around the Digest parameter: Authorization: Digest username="Alice", realm="Bob's Friends", nonce="84a4cc6f3082121f32b42a2187831a9e", response="7587245234b3434cc3412213e5f113a5432" Various Authors [Page 185] Internet Draft SIP January 28, 2002 24.8 Call-ID The Call-ID header field uniquely identifies a particular invitation or all registrations of a particular client. A single multimedia conference can give rise to several calls with different Call-IDs, for example, if a user invites a single individual several times to the same (long-running) conference. Call-IDs are case- sensitive and are simply compared byte-by-byte. The compact form of the Call-ID header field is i. Examples: Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@biloxi.com i:f81d4fae-7dec-11d0-a765-00a0c91e6bf6@192.0.2.4 24.9 Call-Info The Call-Info header field provides additional information about the caller or callee, depending on whether it is found in a request or response. The purpose of the URI is described by the "purpose" parameter. The "icon" parameter designates an image suitable as an iconic representation of the caller or callee. The "info" parameter describes the caller or callee in general, for example, through a web page. The "card" parameter provides a business card, for example, in vCard [33] or LDIF [34] formats. Additonal tokens can be registered using IANA and the procedures in Section 28. Use of the Call-Info header field can pose a security risk. If a callee fetches the URIs provided by a malicious caller, the callee may be at risk for displaying inappropriate or offensive content, dangerous or illegal content, and so on. Therefore, it is RECOMMENDED that a UA only render the information in the Call-Info header if it can verify the authenticity of the element that originated the header and trusts that element. This need not be the peer UA; a proxy can insert this header into requests. Example: Call-Info: ;purpose=icon, ;purpose=info 24.10 Contact Various Authors [Page 186] Internet Draft SIP January 28, 2002 The Contact header field provides a URI whose meaning depends on the the type of request or response it is in. A Contact header field can contain a display name, a URI with URI parameters, and header parameters. This document defines the Contact parameters "q" and "expires". These parameters are only used when the Contact is present in a REGISTER request or response, or in a 3xx response. Additional parameters may be defined in other specifications. When the header field contains a display name, the URI including all URI parameters is enclosed in "<" and ">". If no "<" and ">" are present, all parameters after the URI are header parameters, not URI parameters. The display name can be tokens, or a quoted string, if a larger character set is desired. Even if the "display-name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, semicolon, or question mark. There may or may not be LWS between the display-name and the "<". These rules for parsing a display name, URI and URI parameters, and header parameters also apply for the header fields To and From. The Contact header has a role similar to the Location header field in HTTP. However, the HTTP header field only allows one address, unquoted. Since URIs can contain commas and semicolons as reserved characters, they can be mistaken for header or parameter delimiters, respectively. The compact form of the Contact header field is m (for "moved"). The second example below shows a Contact header field containing both a URI parameter (transport) and a header parameter (expires). Contact: "Mr. Watson" ;q=0.7; expires=3600, "Mr. Watson" ;q=0.1 m: ;expires=60 24.11 Content-Disposition The Content-Disposition header field describes how the message body or, for multipart messages, a message body part is to be interpreted by the UAC or UAS. This SIP header field extends the MIME Content- Type (RFC 1806 [35]). Various Authors [Page 187] Internet Draft SIP January 28, 2002 The value "session" indicates that the body part describes a session, for either calls or early (pre-call) media. The value "render" indicates that the body part should be displayed or otherwise rendered to the user. For backward-compatibility, if the Content- Disposition header is missing, the server SHOULD assume bodies of Content-Type application/sdp are the disposition "session", while other content types are "render". The disposition type "icon" indicates that the body part contains an image suitable as an iconic representation of the caller or callee. The value "alert" indicates that the body part contains information, such as an audio clip, that should be rendered instead of ring tone. The handling parameter, handling-parm, describes how the UAS should react if it receives a message body whose content type or disposition type it does not understand. The parameter has defined values of "optional" and "required". If the handling parameter is missing, the value "required" SHOULD be assumed. If this header field is missing, the MIME type determines the default content disposition. If there is none, "render" is assumed. Example: Content-Disposition: session 24.12 Content-Encoding The Content-Encoding header field is used as a modifier to the "media-type". When present, its value indicates what additional content codings have been applied to the entity-body, and thus what decoding mechanisms MUST be applied in order to obtain the media-type referenced by the Content-Type header field. Content-Encoding is primarily used to allow a body to be compressed without losing the identity of its underlying media type. If multiple encodings have been applied to an entity, the content codings MUST be listed in the order in which they were applied. All content-coding values are case-insensitive. IANA acts as a registry for content-coding value tokens. See [H3.5] for a definition of the syntax for content-coding. Clients MAY apply content encodings to the body in requests. A server MAY apply content encodings to the bodies in responses. The server MUST only use encodings listed in the Accept-Encoding header in the request. Various Authors [Page 188] Internet Draft SIP January 28, 2002 The compact form of the Content-Encoding header field is e. Examples: Content-Encoding: gzip e: tar 24.13 Content-Language See [H14.12]. Example: Content-Language: fr 24.14 Content-Length The Content-Length header field indicates the size of the message- body, in decimal number of octets, sent to the recipient. Applications SHOULD use this field to indicate the size of the message-body to be transferred, regardless of the media type of the entity. If TCP is used as transport, the header field MUST be used. The size of the message-body does not include the CRLF separating headers and body. Any Content-Length greater than or equal to zero is a valid value. If no body is present in a message, then the Content- Length header field MUST be set to zero. The ability to omit Content-Length simplifies the creation of cgi-like scripts that dynamically generate responses. The compact form of the header is l. Examples: Content-Length: 349 l: 173 24.15 Content-Type The Content-Type header field indicates the media type of the message-body sent to the recipient. The "media-type" element is defined in [H3.7]. The Content-Type header MUST be present if the body is not empty. If the body is empty, and a Content-Type header is present, it indicates that the body of the specific type has zero length (for example, an empty audio file). Various Authors [Page 189] Internet Draft SIP January 28, 2002 The compact form of the header is c. Examples: Content-Type: application/sdp c: text/html; charset=ISO-8859-4 24.16 CSeq A CSeq header field in a request contains a single decimal sequence number and the request method. The sequence number MUST be expressible as a 32-bit unsigned integer. The CSeq header serves to order transactions within a dialog, to provide a means to uniquely identify transactions, and to differentiate between new requests and request retransmissions. Example: CSeq: 4711 INVITE 24.17 Date The Date header field contains an RFC 1123 date (see [H14.18]). Unlike HTTP/1.1, SIP only supports the most recent RFC 1123 [36] format for dates. As in [H3.3], SIP restricts the timezone in SIP- date to "GMT", while RFC 1123 allows any timezone. rfc1123-date is case-sensitive. The Date header field reflects the time when the request or response is first sent. The Date header field can be used by simple end systems without a battery-backed clock to acquire a notion of current time. However, in its GMT form, it requires clients to know their offset from GMT. Example: Date: Sat, 13 Nov 2010 23:29:00 GMT Various Authors [Page 190] Internet Draft SIP January 28, 2002 24.18 Error-Info The Error-Info header field provides a pointer to additional information about the error status response. SIP UACs have user interface capabilities ranging from pop-up windows and audio on PC softclients to audio-only on "black" phones or endpoints connected via gateways. Rather than forcing a server generating an error to choose between sending an error status code with a detailed reason phrase and playing an audio recording, the Error-Info header field allows both to be sent. The UAC then has the choice of which error indicator to render to the caller. A UAC MAY treat a SIP URI in an Error-Info header field as if it were a Contact in a redirect and generate a new INVITE, resulting in a recorded announcement session being established. A non-SIP URI MAY be rendered to the user. Examples: SIP/2.0 404 The number you have dialed is not in service Error-Info: 24.19 Expires The Expires header field gives the relative time after which the message (or content) expires. The precise meaning of this is method dependent. The expiration time in an INVITE does not affect the duration of the actual session that may result from the invitation. Session description protocols may offer the ability to express time limits on the session duration, however. The value of this field is an integer number of seconds (in decimal), measured from the receipt of the request. Examples: Expires: 5 24.20 From Various Authors [Page 191] Internet Draft SIP January 28, 2002 The From header field indicates the initiator of the request. This may be different from the initiator of the dialog. Requests sent by the callee to the caller use the callee's address in the From header field. The optional "display-name" is meant to be rendered by a human user interface. A system SHOULD use the display name "Anonymous" if the identity of the client is to remain hidden. Even if the "display- name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, question mark, or semicolon. Syntax issues are discussed in Section 7.3.1. Section 12 describes how From header fields are compared for the purpose of matching requests to dialogs. See Section 24.10 for the rules for parsing a display name, URI and URI parameters, and header parameters. The compact form of the header is f. Examples: From: "A. G. Bell" ;tag=a48s From: sip:+12125551212@server.phone2net.com;tag=887s f: Anonymous ;tag=hyh8 24.21 In-Reply-To The In-Reply-To header field enumerates the Call-IDs that this call references or returns. These Call-IDs may have been cached by the client then included in this header in a return call. This allows automatic call distribution systems to route return calls to the originator of the first call. This also allows callees to filter calls, so that only return calls for calls they originated will be accepted. This field is not a substitute for request authentication. Example: In-Reply-To: 70710@saturn.bell-tel.com, 17320@saturn.bell-tel.com 24.22 Max-Forwards Various Authors [Page 192] Internet Draft SIP January 28, 2002 The Max-Forwards header field must be used with any SIP method to limit the number of proxies or gateways that can forward the request to the next downstream server. This can also be useful when the client is attempting to trace a request chain that appears to be failing or looping in mid-chain. The Max-Forwards value is a decimal integer indicating the remaining number of times this request message is allowed to be forwarded. This count is decremented by each server that forwards the request. This header field should be inserted by elements that can not otherwise guarantee loop detection. For example, a B2BUA should insert a Max-Forwards header field. Example: Max-Forwards: 6 24.23 Min-Expires The Min-Expires header field conveys the minimum registration expiration interval to a registrar. The header field contains a decimal integer number of seconds. The use of the header field in a 423 (Registration Too Brief) response is described in Sections 10.2.8, 10.3, and 25.4.17. Example: Min-Expires: 60 24.24 MIME-Version See [H19.4.1]. Example: MIME-Version: 1.0 24.25 Organization The Organization header field conveys the name of the organization to which the entity issuing the request or response belongs. Various Authors [Page 193] Internet Draft SIP January 28, 2002 The field MAY be used by client software to filter calls. Example: Organization: Boxes by Bob 24.26 Priority The Priority header field indicates the urgency of the request as perceived by the client. The Priority header field describes the priority that the SIP request should have to the receiving human or its agent. For example, it may be factored into decisions about call routing and acceptance. It does not influence the use of communications resources such as packet forwarding priority in routers or access to circuits in PSTN gateways. The header field can have the values "non-urgent", "normal", "urgent", and "emergency", but additional values can be defined elsewhere. It is RECOMMENDED that the value of "emergency" only be used when life, limb, or property are in imminent danger. Otherwise, there are no semantics defined for this header field. These are the values of RFC 2076 [37], with the addition of "emergency". Examples: Subject: A tornado is heading our way! Priority: emergency or Subject: Weekend plans Priority: non-urgent 24.27 Proxy-Authenticate The Proxy-Authenticate header field contains an authentication challenge. The syntax for this header and its use is defined in [H14.33]. See 20.3 for further details on its usage. Example: Various Authors [Page 194] Internet Draft SIP January 28, 2002 Proxy-Authenticate: Digest realm="Carrier SIP", domain="sip:ss1.carrier.com", nonce="f84f1cec41e6cbe5aea9c8e88d359", opaque="", stale=FALSE, algorithm=MD5 24.28 Proxy-Authorization The Proxy-Authorization header field allows the client to identify itself (or its user) to a proxy that requires authentication. The Proxy-Authorization field value consists of credentials containing the authentication information of the user agent for the proxy and/or realm of the resource being requested. See [H14.34] for a definition of the syntax, and section 20.3 for a discussion of its usage. This header field, along with Authorization, breaks the general rules about multiple header fields. Although not a comma-separated list, this header field may be present multiple times, and MUST NOT be combined into a single header using the usual rules described in Section 7.3.1. Example: Proxy-Authorization: Digest username="Alice", realm="Atlanta ISP", nonce="c60f3082ee1212b402a21831ae", response="245f23415f11432b3434341c022" 24.29 Proxy-Require The Proxy-Require header field is used to indicate proxy-sensitive features that must be supported by the proxy. See Section 24.33 for more details on the mechanics of this message and a usage example. Example: Proxy-Require: foo 24.30 RAck The RAck header is sent in a PRACK request to support reliability of provisional responses. It contains two numbers and a method tag. The Various Authors [Page 195] Internet Draft SIP January 28, 2002 first number is the value from the RSeq header in the provisional response that is being acknowledged. The next number, and the method, are copied from the CSeq in the response that is being acknowledged. The method name in the RAck header is case sensitive. Example: RAck: 776656 1 INVITE 24.31 Record-Route The Record-Route is inserted by proxies in a request to force future requests in the session to be routed through the proxy. Details of its use with the Route header field are described in Section 16.4. Example: Record-Route: , 24.32 Reply-To The Reply-To header field contains a logical return URI which may be different from the From header field. For example, the URI MAY be used to return missed calls or unestablished sessions. If the user wished to remain anonymous, the header field SHOULD either be omitted from the request or populated in such as way that does not reveal any private information. Even if the "display-name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, question mark, or semicolon. Syntax issues are discussed in Section 7.3.1. Example: Reply-To: Bob 24.33 Require Various Authors [Page 196] Internet Draft SIP January 28, 2002 The Require header field is used by UACs to tell UASs about options that the UAC expects the UAS to support in order to process the request. Although an optional header, the Require MUST NOT be ignored if it is present. The Require header contains a list of option tags, described in Section 23.2. Each option tag defines a SIP extension that MUST be understood to process the request. Frequently, this is used to indicate that a specific set of extension headers need to be understood. A UAC compliant to this specification MUST only include option tags corresponding to standards-track RFCs. Example: Require: 100rel 24.34 Retry-After The Retry-After header field can be used with a 503 (Service Unavailable) response to indicate how long the service is expected to be unavailable to the requesting client and with a 404 (Not Found), 600 (Busy), or 603 (Decline) response to indicate when the called party anticipates being available again. The value of this field is a positive integer number of seconds (in decimal) after the time of the response. An optional comment can be used to indicate additional information about the time of callback. An optional "duration" parameter indicates how long the called party will be reachable starting at the initial time of availability. If no duration parameter is given, the service is assumed to be available indefinitely. Examples: Retry-After: 18000;duration=3600 Retry-After: 120 (I'm in a meeting) 24.35 Route The Route is used to force routing for a request through the listed set of proxies. Details of its use with the Record-Route header field are described in Section 13. Various Authors [Page 197] Internet Draft SIP January 28, 2002 Example: Route: , 24.36 RSeq The RSeq header is used in provisional responses in order to transmit them reliably. It contains a single numeric value from 1 to 2**32 - 1. For details on its usage, see Section 18.1. Example: RSeq: 988789 24.37 Server The Server header field contains information about the software used by the UAS to handle the request. The syntax for this field is defined in [H14.38]. Revealing the specific software version of the server might allow the server to become more vulnerable to attacks against software that is known to contain security holes. Implementors SHOULD make the Server header field a configurable option. Example: Server: HomeProxy v2 24.38 Subject The Subject header field provides a summary or indicates the nature of the call, allowing call filtering without having to parse the session description. The session description does not have to use the same subject indication as the invitation. The compact form of the header is s. Example: Subject: Need more boxes Various Authors [Page 198] Internet Draft SIP January 28, 2002 s: Tech Support 24.39 Supported The Supported header field enumerates all the extensions supported by the UAC or UAS. The Supported header contains a list of option tags, described in Section 23.2, that are understood by the UAC or UAS. A UA compliant to this specification MUST only include option tags corresponding to standards-track RFCs. If empty, it means that no extensions are supported. Example: Supported: 100rel 24.40 Timestamp The Timestamp header field describes when the UAC sent the request to the UAS. See Section 8.2.6 for details on how to generate a response to a request that contains the header field, and Section 17.3 for usage in RTT estimation. Example: Timestamp: 54 24.41 To The To header field specifies the logical recipient of the request. The optional "display-name" is meant to be rendered by a human-user interface. The "tag" parameter serves as a general mechanism to distinguish multiple instances of a user identified by a single SIP URI. See Section 13 for details of the "tag" parameter. Section 12 describes how To and From header fields are compared for the purpose of matching requests to dialogs. See Section 24.10 for the rules for parsing a display name, URI and URI parameters, and Various Authors [Page 199] Internet Draft SIP January 28, 2002 header parameters. The compact form of the header is t. The following are examples of valid To headers: To: The Operator ;tag=287447 t: sip:+12125551212@server.phone2net.com 24.42 Unsupported The Unsupported header field lists the features not supported by the UAS. See Section 24.33 for motivation. Example: Unsupported: foo 24.43 User-Agent The User-Agent header field contains information about the UAC originating the request. The syntax and semantics are defined in [H14.43]. Revealing the specific software version of the user agent might allow the user agent to become more vulnerable to attacks against software that is known to contain security holes. Implementors SHOULD make the User-Agent header field a configurable option. Example: User-Agent: Softphone Beta1.5 24.44 Via The Via field indicates the path taken by the request so far and indicates the path that should be followed in routing responses. The branch ID parameter in the Via header serves as a transaction identifier, and is used by proxies to detect loops. The Via header field contains the transport protocol used to send the message, the client's host name or network address and, if not the Various Authors [Page 200] Internet Draft SIP January 28, 2002 default port number, the port number at which it wishes to receive responses. The Via header field can also contain parameters such as "maddr", "ttl", "received", and "branch", whose meaning and use are described in other sections. Transport protocols defined here are "UDP", "TCP", "TLS", and "SCTP". "TLS" means TLS over TCP. The host or network address and port number are not required to follow the SIP URI syntax. Specifically, LWS on either side of the ":" or "/" is allowed, as shown in the second example below. Via: SIP/2.0/UDP erlang.bell-telephone.com:5060;branch=z9hG4bK87asdks7 Via: SIP/2.0/UDP 128.59.16.1:5060 ;received=128.59.19.3;branch=z9hG4bK77asjd The compact form of the header is v. In this example, the message originated from a multi-homed host with two addresses, 128.59.16.1 and 128.59.19.3. The sender guessed wrong as to which network interface would be used. Erlang.bell- telephone.com noticed the mismatch and added a parameter to the previous hop's Via header field, containing the address that the packet actually came from. Another example: Via: SIP / 2.0 / UDP first.example.com: 4000;ttl=16 ;maddr=224.2.0.1 ;branch=z9hG4bKa7c6a8dlze.1 Even though this specification mandates that the branch parameter be present in all requests, the BNF for the header indicates that it is optional. This allows interoperation with RFC 2543 elements, which did not have to insert the branch parameter. 24.45 Warning The Warning header field is used to carry additional information about the status of a response. Warning headers are sent with responses and contain a three-digit warning code, host name, and warning text. The "warn-text" should be in a natural language that is most likely to be intelligible to the human user receiving the response. This Various Authors [Page 201] Internet Draft SIP January 28, 2002 decision can be based on any available knowledge, such as the location of the user, the Accept-Language field in a request, or the Content-Language field in a response. The default language is i- default [38]. The currently-defined "warn-code"s are listed below, with a recommended warn-text in English and a description of their meaning. These warnings describe failures induced by the session description. The first digit of warning codes beginning with "3" indicates warnings specific to SIP. Warnings 300 through 329 are reserved for indicating problems with keywords in the session description, 330 through 339 are warnings related to basic network services requested in the session description, 370 through 379 are warnings related to quantitative QoS parameters requested in the session description, and 390 through 399 are miscellaneous warnings that do not fall into one of the above categories. 300 Incompatible network protocol: One or more network protocols contained in the session description are not available. 301 Incompatible network address formats: One or more network address formats contained in the session description are not available. 302 Incompatible transport protocol: One or more transport protocols described in the session description are not available. 303 Incompatible bandwidth units: One or more bandwidth measurement units contained in the session description were not understood. 304 Media type not available: One or more media types contained in the session description are not available. 305 Incompatible media format: One or more media formats contained in the session description are not available. 306 Attribute not understood: One or more of the media attributes in the session description are not supported. 307 Session description parameter not understood: A parameter other than those listed above was not understood. 330 Multicast not available: The site where the user is located does not support multicast. 331 Unicast not available: The site where the user is located Various Authors [Page 202] Internet Draft SIP January 28, 2002 does not support unicast communication (usually due to the presence of a firewall). 370 Insufficient bandwidth: The bandwidth specified in the session description or defined by the media exceeds that known to be available. 399 Miscellaneous warning: The warning text can include arbitrary information to be presented to a human user or logged. A system receiving this warning MUST NOT take any automated action. 1xx and 2xx have been taken by HTTP/1.1. Additional "warn-code"s, as in the example below, can be defined through IANA. Examples: Warning: 307 isi.edu "Session parameter 'foo' not understood" Warning: 301 isi.edu "Incompatible network address type 'E.164'" 24.46 WWW-Authenticate The WWW-Authenticate header field contains an authentication challenge. The syntax for this header field and use is defined in [H14.47]. See 20.2 for further details on its usage. Example: WWW-Authenticate: Digest realm="Bob's Friends", domain="sip:boxesbybob.com", nonce="f84f1cec41e6cbe5aea9c8e88d359", opaque="", stale=FALSE, algorithm=MD5 25 Response Codes The response codes are consistent with, and extend, HTTP/1.1 response codes. Not all HTTP/1.1 response codes are appropriate, and only those that are appropriate are given here. Other HTTP/1.1 response codes SHOULD NOT be used. Response codes not defined by HTTP/1.1 have codes x80 upwards to avoid clashes with future HTTP response codes. Various Authors [Page 203] Internet Draft SIP January 28, 2002 Also, SIP defines a new class, 6xx. 25.1 Provisional 1xx Provisional responses, also known as informational responses, indicate that the server or proxy contacted is performing some further action and does not yet have a definitive response. A server typically sends a 1xx response if it expects to take more than 200 ms to obtain a final response. Note that 1xx responses are not transmitted reliably, that is, they do not cause the client to send an ACK. Provisional (1xx) responses MAY contain message bodies, including session descriptions. 25.1.1 100 Trying This response indicates that the request has been received by the next hop server and that some unspecified action is being taken on behalf of this call (e.g., a database is being consulted). This response, like all other provisional responses, stops retransmissions of an INVITE by a UAC. The 100 (Trying) response is different from other provisional responses, in that it is never forwarded upstream by a stateful proxy. 25.1.2 180 Ringing The user agent receiving the INVITE is trying to alert the user. This response MAY be used to initiate local ringback. 25.1.3 181 Call Is Being Forwarded A proxy server MAY use this status code to indicate that the call is being forwarded to a different set of destinations. 25.1.4 182 Queued The called party is temporarily unavailable, but the callee has decided to queue the call rather than reject it. When the callee becomes available, it will return the appropriate final status response. The reason phrase MAY give further details about the status of the call, e.g., "5 calls queued; expected waiting time is 15 minutes". The server MAY issue several 182 (Queued) responses to update the caller about the status of the queued call. 25.1.5 183 Session Progress The 183 (Session Progress) response is used to convey information about the progress of the call which is not otherwise classified. The Reason-Phrase, header fields, or message body MAY be used to convey Various Authors [Page 204] Internet Draft SIP January 28, 2002 more details about the call progress. 25.2 Successful 2xx The request was successful. 25.2.1 200 OK The request has succeeded. The information returned with the response depends on the method used in the request. 25.3 Redirection 3xx 3xx responses give information about the user's new location, or about alternative services that might be able to satisfy the call. 25.3.1 300 Multiple Choices The address in the request resolved to several choices, each with its own specific location, and the user (or user agent) can select a preferred communication end point and redirect its request to that location. The response MAY include a message body containing a list of resource characteristics and location(s) from which the user or user agent can choose the one most appropriate, if allowed by the Accept request header. However, no MIME types have been defined for this message body. The choices SHOULD also be listed as Contact fields (Section 24.10). Unlike HTTP, the SIP response MAY contain several Contact fields or a list of addresses in a Contact field. User agents MAY use the Contact header field value for automatic redirection or MAY ask the user to confirm a choice. However, this specification does not define any standard for such automatic selection. This status response is appropriate if the callee can be reached at several different locations and the server cannot or prefers not to proxy the request. 25.3.2 301 Moved Permanently The user can no longer be found at the address in the Request-URI and the requesting client SHOULD retry at the new address given by the Contact header field (Section 24.10). The requestor SHOULD update any local directories, address books and user location caches with this new value and redirect future requests to the address(es) Various Authors [Page 205] Internet Draft SIP January 28, 2002 listed. 25.3.3 302 Moved Temporarily The requesting client SHOULD retry the request at the new address(es) given by the Contact header field (Section 24.10). The Request-URI of the new request uses the value of the Contact header in the response. The duration of the validity of the Contact URI can be indicated through an Expires (Section 24.19) header field or an expires parameter in the Contact header field. Both proxies and UAs MAY cache this URI for the duration of the expiration time. If there is no explicit expiration time, the address is only valid once for recursing, and MUST NOT be cached for future transactions. If the URI cached from the Contact header field fails, the Request- URI from the redirected request MAY be tried again a single time. The temporary URI may have become out of date sooner than the expiration time, and a new temporary URI may be available. 25.3.4 305 Use Proxy The requested resource MUST be accessed through the proxy given by the Contact field. The Contact field gives the URI of the proxy. The recipient is expected to repeat this single request via the proxy. 305 (Use Proxy) responses MUST only be generated by user agent servers. 25.3.5 380 Alternative Service The call was not successful, but alternative services are possible. The alternative services are described in the message body of the response. Formats for such bodies are not defined here, and may be the subject of future standardization. 25.4 Request Failure 4xx 4xx responses are definite failure responses from a particular server. The client SHOULD NOT retry the same request without modification (e.g., adding appropriate authorization). However, the same request to a different server might be successful. 25.4.1 400 Bad Request Various Authors [Page 206] Internet Draft SIP January 28, 2002 The request could not be understood due to malformed syntax. The Reason-Phrase SHOULD identify the syntax problem in more detail, e.g., "Missing Call-ID header". 25.4.2 401 Unauthorized The request requires user authentication. This response is issued by user agent servers and registrars, while 407 (Proxy Authentication Required) is used by proxy servers. 25.4.3 402 Payment Required Reserved for future use. 25.4.4 403 Forbidden The server understood the request, but is refusing to fulfill it. Authorization will not help, and the request SHOULD NOT be repeated. 25.4.5 404 Not Found The server has definitive information that the user does not exist at the domain specified in the Request-URI. This status is also returned if the domain in the Request-URI does not match any of the domains handled by the recipient of the request. 25.4.6 405 Method Not Allowed The method specified in the Request-Line is understood, but not allowed for the address identified by the Request-URI. The response MUST include an Allow header field containing a list of valid methods for the indicated address. 25.4.7 406 Not Acceptable The resource identified by the request is only capable of generating response entities which have content characteristics not acceptable according to the accept headers sent in the request. 25.4.8 407 Proxy Authentication Required This code is similar to 401 (Unauthorized), but indicates that the client MUST first authenticate itself with the proxy. SIP access authentication is explained in section 22 and 20.3. This status code can be used for applications where access to the communication channel (e.g., a telephony gateway) rather than the callee requires authentication. Various Authors [Page 207] Internet Draft SIP January 28, 2002 25.4.9 408 Request Timeout The server could not produce a response within a suitable amount of time, for example, if it could not determine the location of the user in time. The client MAY repeat the request without modifications at any later time. 25.4.10 410 Gone The requested resource is no longer available at the server and no forwarding address is known. This condition is expected to be considered permanent. If the server does not know, or has no facility to determine, whether or not the condition is permanent, the status code 404 (Not Found) SHOULD be used instead. 25.4.11 413 Request Entity Too Large The server is refusing to process a request because the request entity is larger than the server is willing or able to process. The server MAY close the connection to prevent the client from continuing the request. If the condition is temporary, the server SHOULD include a Retry- After header field to indicate that it is temporary and after what time the client MAY try again. 25.4.12 414 Request-URI Too Long The server is refusing to service the request because the Request-URI is longer than the server is willing to interpret. 25.4.13 415 Unsupported Media Type The server is refusing to service the request because the message body of the request is in a format not supported by the server for the requested method. The server SHOULD return a list of acceptable formats using the Accept, Accept-Encoding and Accept-Language header fields. UAC processing of this response is described in Section 8.1.4.6. 25.4.14 416 Unsupported URI Scheme The server cannot process the request because the scheme of the URI in the Request-URI is unknown to the server. Client processing of this response is described in Section 8.1.4.6. 25.4.15 420 Bad Extension Various Authors [Page 208] Internet Draft SIP January 28, 2002 The server did not understand the protocol extension specified in a Proxy-Require (Section 24.29) or Require (Section 24.33) header field. The server SHOULD include a list of the unsupported extensions in an Unsupported header in the response. UAC processing of this response is described in Section 8.1.4.6. 25.4.16 421 Extension Required The UAS needs a particular extension to process the request, but this extension is not listed in a Supported header in the request. Responses with this status code MUST contain a Require header field listing the required extensions. A UAS SHOULD NOT use this response unless it truly cannot provide any useful service to the client. Instead, if a desirable extension is not listed in the Supported header field, servers SHOULD process the request using baseline SIP capabilities and any extensions supported by the client. 25.4.17 423 Registration Too Brief The registrar is rejecting a registration request because a Contact header field expiration time was too small. The use of this response and the related Min-Expires header field are described in Sections 10.2.8, 10.3, and 24.23. 25.4.18 480 Temporarily Unavailable The callee's end system was contacted successfully but the callee is currently unavailable (e.g., is not logged in, logged in in such a manner as to preclude communication with the callee or has activated the "do not disturb" feature). The response MAY indicate a better time to call in the Retry-After header. The user could also be available elsewhere (unbeknownst to this host). The reason phrase SHOULD indicate a more precise cause as to why the callee is unavailable. This value SHOULD be setable by the user agent. Status 486 (Busy Here) MAY be used to more precisely indicate a particular reason for the call failure. This status is also returned by a redirect or proxy server that recognizes the user identified by the Request-URI, but does not currently have a valid forwarding location for that user. 25.4.19 481 Call/Transaction Does Not Exist This status indicates that the UAS received a request that does not match any existing dialog or transaction. Various Authors [Page 209] Internet Draft SIP January 28, 2002 25.4.20 482 Loop Detected The server has detected a loop (Section 2). 25.4.21 483 Too Many Hops The server received a request that contains a Max-Forwards (Section 24.22) header with the value zero. 25.4.22 484 Address Incomplete The server received a request with a Request-URI that was incomplete. Additional information SHOULD be provided in the reason phrase. This status code allows overlapped dialing. With overlapped dialing, the client does not know the length of the dialing string. It sends strings of increasing lengths, prompting the user for more input, until it no longer receives a 484 (Address Incomplete) status response. 25.4.23 485 Ambiguous The Request-URI was ambiguous. The response MAY contain a listing of possible unambiguous addresses in Contact header fields. Revealing alternatives can infringe on privacy of the user or the organization. It MUST be possible to configure a server to respond with status 404 (Not Found) or to suppress the listing of possible choices for ambiguous Request-URIs. Example response to a request with the Request-URI sip:lee@example.com : 485 Ambiguous SIP/2.0 Contact: Carol Lee Contact: Ping Lee Contact: Lee M. Foote Some email and voice mail systems provide this functionality. A status code separate from 3xx is used since the semantics are different: for 300, it is assumed that the same person or service will be reached by the choices provided. While an automated choice or sequential search makes sense for a 3xx response, user intervention is required for a 485 (Ambiguous) response. Various Authors [Page 210] Internet Draft SIP January 28, 2002 25.4.24 486 Busy Here The callee's end system was contacted successfully but the callee is currently not willing or able to take additional calls at this end system. The response MAY indicate a better time to call in the Retry-After header. The user could also be available elsewhere, such as through a voice mail service. Status 600 (Busy Everywhere) SHOULD be used if the client knows that no other end system will be able to accept this call. 25.4.25 487 Request Terminated The request was terminated by a BYE or CANCEL request. This response is never returned for a CANCEL request itself. 25.4.26 488 Not Acceptable Here The response has the same meaning as 606 (Not Acceptable), but only applies to the specific entity addressed by the Request-URI and the request may succeed elsewhere. A message body containing a description of media capabilities MAY be present in the response, which is formatted according to the Accept header field in the INVITE (or application/sdp if not present), the same as a message body in a 200 (OK) response to an OPTIONS request. 25.4.27 491 Request Pending The request was received by a UAS which had a pending request within the same dialog. Section 14.2 describes how such "glare" situations are resolved. 25.4.28 493 Undecipherable The request was received by a UAS which contained an encrypted MIME body for which the recipient does not possess or will not provide an appropriate decryption key. This response MAY have a single body containing an appropriate public key that should be used to encrypt MIME bodies sent to this user agent. Details of the usage of this response codecan be found in Section 21.2. 25.5 Server Failure 5xx 5xx responses are failure responses given when a server itself has erred. 25.5.1 500 Server Internal Error The server encountered an unexpected condition that prevented it from Various Authors [Page 211] Internet Draft SIP January 28, 2002 fulfilling the request. The client MAY display the specific error condition, and MAY retry the request after several seconds. If the condition is temporary, the server MAY indicate when the client may retry the request using the Retry-After header. 25.5.2 501 Not Implemented The server does not support the functionality required to fulfill the request. This is the appropriate response when a UAS does not recognize the request method and is not capable of supporting it for any user. (Proxies forward all requests regardless of method.) Note that a 405 (Method Not Allowed) is sent when the server recognizes the request method, but that method is not allowed or supported. 25.5.3 502 Bad Gateway The server, while acting as a gateway or proxy, received an invalid response from the downstream server it accessed in attempting to fulfill the request. 25.5.4 503 Service Unavailable The server is temporarily unable to process the request due to a temporary overloading or maintenance of the server. The server MAY indicate when the client should retry the request in a Retry-After header. If no Retry-After is given, the client MUST act as if it had received a 500 (Server Internal Error) response. A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD attempt to forward the request to an alternate server. It SHOULD NOT forward any other requests to that server for the duration specified in the Retry-After header field, if present. Servers MAY refuse the connection or drop the request instead of responding with 503 (Service Unavailable). 25.5.5 504 Server Time-out The server did not receive a timely response from an external server it accessed in attempting to process the request. 408 (Request Timeout) should be used instead if there was no response within the period specified in the Expires header field from the upstream server. 25.5.6 505 Version Not Supported The server does not support, or refuses to support, the SIP protocol Various Authors [Page 212] Internet Draft SIP January 28, 2002 version that was used in the request. The server is indicating that it is unable or unwilling to complete the request using the same major version as the client, other than with this error message. 25.5.7 513 Message Too Large The server was unable to process the request since the message length exceeded its capabilities. 25.6 Global Failures 6xx 6xx responses indicate that a server has definitive information about a particular user, not just the particular instance indicated in the Request-URI. 25.6.1 600 Busy Everywhere The callee's end system was contacted successfully but the callee is busy and does not wish to take the call at this time. The response MAY indicate a better time to call in the Retry-After header. If the callee does not wish to reveal the reason for declining the call, the callee uses status code 603 (Decline) instead. This status response is returned only if the client knows that no other end point (such as a voice mail system) will answer the request. Otherwise, 486 (Busy Here) should be returned. 25.6.2 603 Decline The callee's machine was successfully contacted but the user explicitly does not wish to or cannot participate. The response MAY indicate a better time to call in the Retry-After header. This status response is returned only if the client knows that no other end point will answer the request. 25.6.3 604 Does Not Exist Anywhere The server has authoritative information that the user indicated in the Request-URI does not exist anywhere. 25.6.4 606 Not Acceptable The user's agent was contacted successfully but some aspects of the session description such as the requested media, bandwidth, or addressing style were not acceptable. A 606 (Not Acceptable) response means that the user wishes to communicate, but cannot adequately support the session described. The 606 (Not Acceptable) response MAY contain a list of reasons in a Various Authors [Page 213] Internet Draft SIP January 28, 2002 Warning header field describing why the session described cannot be supported. A message body containing a description of media capabilities MAY be present in the response, which is formatted according to the Accept header field in the INVITE (or application/sdp if not present), the same as a message body in a 200 (OK) response to an OPTIONS request. Reasons are listed in Section 24.45. It is hoped that negotiation will not frequently be needed, and when a new user is being invited to join an already existing conference, negotiation may not be possible. It is up to the invitation initiator to decide whether or not to act on a 606 (Not Acceptable) response. This status response is returned only if the client knows that no other end point will answer the request. 26 Examples In the following examples, we often omit the message body and the corresponding Content-Length and Content-Type headers for brevity. 26.1 Registration Bob registers on start-up. The message flow is shown in Figure 9. biloxi.com Bob's registrar softphone | | | REGISTER F1 | |<---------------| | 200 OK F2 | |--------------->| Figure 9: SIP Registration Example F1 REGISTER Bob -> Registrar REGISTER sip:registrar.biloxi.com SIP/2.0 Via: SIP/2.0/UDP 192.0.2.4:5060;branch=z9hG4bKnashds7 To: Bob From: Bob ;tag=456248 Call-ID: 843817637684230@998sdasdh09 Various Authors [Page 214] Internet Draft SIP January 28, 2002 CSeq: 1826 REGISTER Contact: Expires: 7200 Content-Length: 0 The registration expires after two hours. The registrar responds with a 200 OK: F2 200 OK Registrar -> Bob SIP/2.0 200 OK Via: SIP/2.0/UDP 192.0.2.4:5060;branch=z9hG4bKnashds7 To: Bob From: Bob ;tag=456248 Call-ID: 843817637684230@998sdasdh09 CSeq: 1826 REGISTER Contact: Expires: 7200 Content-Length: 0 26.2 Session Setup This example contains the full details of the example session setup in Section 4. The message flow is shown in Figure 1. F1 INVITE Alice -> atlanta.com proxy INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: Content-Type: application/sdp Content-Length: 142 (Alice's SDP not shown) Various Authors [Page 215] Internet Draft SIP January 28, 2002 F2 100 Trying atlanta.com proxy -> Alice SIP/2.0 100 Trying Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Content-Length: 0 F3 INVITE atlanta.com proxy -> biloxi.com proxy INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: Content-Type: application/sdp Content-Length: 142 (Alice's SDP not shown) F4 100 Trying biloxi.com proxy -> atlanta.com proxy SIP/2.0 100 Trying Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Content-Length: 0 Various Authors [Page 216] Internet Draft SIP January 28, 2002 F5 INVITE biloxi.com proxy -> Bob INVITE sip:bob@192.0.2.4 SIP/2.0 Via: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1 Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: Content-Type: application/sdp Content-Length: 142 (Alice's SDP not shown) F6 180 Ringing Bob -> biloxi.com proxy SIP/2.0 180 Ringing Via: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1 Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob ;tag=a6c85cf From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Content-Length: 0 F7 180 Ringing biloxi.com proxy -> atlanta.com proxy SIP/2.0 180 Ringing Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob ;tag=a6c85cf From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Content-Length: 0 Various Authors [Page 217] Internet Draft SIP January 28, 2002 F8 180 Ringing atlanta.com proxy -> Alice SIP/2.0 180 Ringing Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob ;tag=a6c85cf From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Content-Length: 0 F9 200 OK Bob -> biloxi.com proxy SIP/2.0 200 OK Via: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1 Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob ;tag=a6c85cf From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: Content-Type: application/sdp Content-Length: 131 (Bob's SDP not shown) F10 200 OK biloxi.com proxy -> atlanta.com proxy SIP/2.0 200 OK Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob ;tag=a6c85cf From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: Content-Type: application/sdp Content-Length: 131 (Bob's SDP not shown) Various Authors [Page 218] Internet Draft SIP January 28, 2002 F11 200 OK atlanta.com proxy -> Alice SIP/2.0 200 OK Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 To: Bob ;tag=a6c85cf From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: Content-Type: application/sdp Content-Length: 131 (Bob's SDP not shown) F12 ACK Alice -> Bob ACK sip:bob@192.0.2.4 SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds9 To: Bob ;tag=a6c85cf From: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 ACK Content-Length: 0 The media session between Alice and Bob is now established. Bob hangs up first. Note that Bob's SIP phone maintains its own CSeq numbering space, which, in this example, begins with 231. Since Bob is making the request, the To and From URIs and tags have been swapped. F13 BYE Bob -> Alice BYE sip:alice@pc33.atlanta.com SIP/2.0 Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKnashds10 From: Bob ;tag=a6c85cf To: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 231 BYE Content-Length: 0 Various Authors [Page 219] Internet Draft SIP January 28, 2002 F14 200 OK Alice -> Bob SIP/2.0 200 OK Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKnashds10 From: Bob ;tag=a6c85cf To: Alice ;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 231 BYE Content-Length: 0 The SIP Call Flows document [39] contains further examples of SIP messages. ;; This buffer is for notes you don't want to save, and for Lisp evaluation. ;; If you want to create a file, first visit that file with C-x C-f, ;; then enter the text in that file's own buffer. 27 Augmented BNF for the SIP Protocol All of the mechanisms specified in this document are described in both prose and an augmented Backus-Naur Form (BNF) similar to that used by RFC 2234 [40]. Implementors need to be familiar with the notation in order to understand this specification. The augmented BNF includes the following constructs: name = definition The name of a rule is simply the name itself (without any enclosing "<" and ">") and is separated from its definition by the equal "=" character. White space is only significant in that the indentation of continuation lines indicates a rule definition that spans more than one line. Certain basic rules are in uppercase, such as SP, LWS, HT, CRLF, DIGIT, ALPHA, etc. Angle brackets are used within definitions to clarify the use of rule names. "literal" Quotation marks surround literal text. Unless stated otherwise, the text is case-insensitive. Various Authors [Page 220] Internet Draft SIP January 28, 2002 rule1 | rule2 Elements separated by a bar ("|") are alternatives, that is, "yes | no" will accept yes or no. (rule1 rule2) Elements enclosed in parentheses are treated as a single element. Thus, "(elem (foo | bar) elem)" allows the token sequences "elem foo elem" and "elem bar elem". *rule The character "*" preceding an element indicates repetition. The full form is "*element" indicating at least and at most occurrences of element. Default values are 0 and infinity so that "*(element)" allows any number, including zero; "1*element" requires at least one; and "1*2element" allows one or two. [rule] Square brackets enclose optional elements; "[foo bar]" is equivalent to "*1(foo bar)". N rule Specific repetition: "(element)" is equivalent to "*(element)"; that is, exactly occurrences of (element). Thus 2DIGIT is a 2-digit number, and 3ALPHA is a string of three alphabetic characters. ; comment A semi-colon, set off some distance to the right of rule text, starts a comment that continues to the end of line. This is a simple way of including useful notes in parallel with the specifications. Various Authors [Page 221] Internet Draft SIP January 28, 2002 27.1 Basic Rules The following rules are used throughout this specification to describe basic parsing constructs. The US-ASCII coded character set is defined by ANSI X3.4-1986. OCTET = %x00-ff ; any 8-bit sequence of data CHAR = %x00-7f ; any US-ASCII character (octets 0 - 127) upalpha = "A" | "B" | "C" | "D" | "E" | "F" | "G" | "H" | "I" | "J" | "K" | "L" | "M" | "N" | "O" | "P" | "Q" | "R" | "S" | "T" | "U" | "V" | "W" | "X" | "Y" | "Z" lowalpha = "a" | "b" | "c" | "d" | "e" | "f" | "g" | "h" | "i" | "j" | "k" | "l" | "m" | "n" | "o" | "p" | "q" | "r" | "s" | "t" | "u" | "v" | "w" | "x" | "y" | "z" alpha = lowalpha | upalpha DIGIT = "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" | "8" | "9" alphanum = alpha | DIGIT CTL = %x00-1f | %x7f ; (octets 0 -- 31) and DEL (127) CR = %d13 ; US-ASCII CR, carriage return character LF = %d10 ; US-ASCII LF, line feed character SP = %d32 ; US-ASCII SP, space character HT = %d09 ; US-ASCII HT, horizontal tab character CRLF = CR LF ; typically the end of a line The following are defined in RFC 2396 [9] for the SIP URI: reserved = ";" | "/" | "?" | ":" | "@" | " " | "" | "+" | "$" | "," unreserved = alphanum | mark mark = "-" | "_" | "." | "!" | "~" | "*" | "'" |"(" | ")" escaped = "%" hex hex SIP header field values can be folded onto multiple lines if the continuation line begins with a space or horizontal tab. All linear white space, including folding, has the same semantics as SP. A recipient MAY replace any linear white space with a single SP before interpreting the field value or forwarding the message downstream. This is intended to behave exactly as HTTP 1.1 as described in RFC2615 [12]. The SWS construct is similar to LWS but allows zero instances of space or tab Various Authors [Page 222] Internet Draft SIP January 28, 2002 LWS = *( SP | HT ) [CRLF] 1*( SP | HT ) ; linear whitespace SWS = *( SP | HT ) [CRLF] *( SP | HT ) ; sep whitespace To separate the header name from the rest of value, a colon is used, which, by the above rule, allows whitespace before, but no line break, and whitespace after, including a linebreak. The HCOLON defines this construct. HCOLON = *( SP | HT ) ":" SWS The TEXT-UTF8 rule is only used for descriptive field contents and values that are not intended to be interpreted by the message parser. Words of *TEXT-UTF8 contain characters from the UTF-8 character set (RFC 2279 [11]). The TEXT-UTF8-TRIM rule is used for descriptive field contents that are not quoted strings, where leading and trailing LWS is not meaningful. In this regard, SIP differs from HTTP, which uses the ISO 8859-1 character set. TEXT-UTF8 = *(TEXT-UTF8char | LWS) TEXT-UTF8-TRIM = *TEXT-UTF8char *(*LWS TEXT-UTF8char) TEXT-UTF8char = %x21-7e | UTF8-NONASCII UTF8-NONASCII = %xc0-df 1UTF8-CONT | %xe0-ef 2UTF8-CONT | %xf0-f7 3UTF8-CONT | %xf8-fb 4UTF8-CONT | %xfc-fd 5UTF8-CONT UTF8-CONT = %x80-bf A CRLF is allowed in the definition of TEXT-UTF8 only as part of a header field continuation. It is expected that the folding LWS will be replaced with a single SP before interpretation of the TEXT-UTF8 value. Hexadecimal numeric characters are used in several protocol elements. Some elements (authentication) force hex alphas to be lower case. LHEX = digit | "a" | "b" | "c" | "d" | "e" | "f" Others allow mixed upper and lower case Various Authors [Page 223] Internet Draft SIP January 28, 2002 hex = LHEX | "A" | "B" | "C" | "D" | "E" | "F" Many SIP header field values consist of words separated by LWS or special characters. Unless otherwise stated, tokens are case- insensitive. These special characters MUST be in a quoted string to be used within a parameter value. The word construct is used in Call-ID to allow most separators to be used. token = 1*(alphanum | "-" | "." | "!" | "%" | "*" | "_" | "+" | "`" | "'" | "~" ) separators = "(" | ")" | "<" | ">" | "@" | "," | ";" | ":" | "\" | <"> | "/" | "[" | "]" | "?" | "=" | "{" | "}" | SP | HT word = 1*(alphanum | "-" | "." | "!" | "%" | "*" | "_" | "+" | "`" | "'" | "~" "(" | ")" | "<" | ">" ":" | "\" | <"> | "/" | "[" | "]" | "?" | "{" | "}" | SP | HT ) When tokens are used or separators are used between elements, whitespace is often allowed before or after these characters: MINUS = SWS "-" SWS ; minus DOT = SWS "." SWS ; period PERCENT = SWS "%" SWS ; percent BANG = SWS "!" SWS ; exclamation PLUS = SWS "+" SWS ; plus STAR = SWS "*" SWS ; asterisk SLASH = SWS "/" SWS ; slash TILDE = SWS "~" SWS ; tilde EQUAL = SWS "=" SWS ; equal LPAREN = SWS "(" SWS ; left parenthesis RPAREN = SWS ")" SWS ; right parenthesis LANGLE = SWS "<" SWS ; left angle bracket RAQUOT = ">" SWS ; right angle quote LAQUOT = SWS "<"; left angle quote RANGLE = SWS ">" SWS ; right angle bracket BAR = SWS "|" SWS ; vertical bar ATSIGN = SWS "@" SWS ; atsign COMMA = SWS "," SWS ; comma Various Authors [Page 224] Internet Draft SIP January 28, 2002 SEMI = SWS ";" SWS ; semicolon COLON = SWS ":" SWS ; colon DQUOT = SWS <"> SWS ; double quotation mark LDQUOT = SWS <">; open double quotation mark RDQUOT = <"> SWS ; close double quotation mark LBRACK = SWS "{" SWS ; left square bracket RBRACK = SWS "}" SWS ; right square bracket Comments can be included in some SIP header fields by surrounding the comment text with parentheses. Comments are only allowed in fields containing "comment" as part of their field value definition. In all other fields, parentheses are considered part of the field value. comment = LPAREN *(ctext | quoted-pair | comment) RPAREN ; ctext includes all chars except left and right parens and backslash ctext = %x21-27 | %x2a-5b | %x5d-7e | UTF8-NONASCII | LWS A string of text is parsed as a single word if it is quoted using double-quote marks. In quoted strings, quotation marks (") and backslashes (\) need to be escaped. quoted-string = ( SWS <"> *(qdtext | quoted-pair ) <"> ) qdtext = LWS | %x21 | %x23-5b | %x5d-7e | UTF8-NONASCII The backslash character ("\") MAY be used as a single-character quoting mechanism only within quoted-string and comment constructs. Unlike HTTP/1.1, the characters CR and LF cannot be escaped by this mechanism to avoid conflict with line folding and header separation. quoted-pair = "\" (%x00 - %x09 | %x0b | %x0c | %x0e - %x7f) SIP-URI = "sip:" [ userinfo "@" ] hostport url-parameters [ headers ] Various Authors [Page 225] Internet Draft SIP January 28, 2002 userinfo = [ user | telephone-subscriber [ ":" password ]] user = *( unreserved | escaped | user-unreserved ) user-unreserved = " " | "=" | "+" | "$" | "," | ";" | "?" | "/" telephone-subscriber __ ["+"] 1*(DIGIT | "-" | ".") password = *( unreserved | escaped | " " | "=" | "+" | "$" | "," ) hostport = host [ ":" port ] host = hostname | IPv4address | IPv6reference hostname = *( domainlabel "." ) toplabel [ "." ] domainlabel = alphanum | alphanum *( alphanum | "-" ) alphanum toplabel = alpha | alpha *( alphanum | "-" ) alphanum IPv4address = 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT IPv6reference = "[" IPv6address "]" IPv6address = hexpart [ ":" IPv4address ] hexpart = hexseq | hexseq "::" [ hexseq ] | "::" [ hexseq ] hexseq = hex4 *( ":" hex4) hex4 = 1*4HEX port = 1*DIGIT url-parameters = *( ";" url-parameter) url-parameter = transport-param | user-param | method-param |ttl-param | maddr-param | other-param transport-param = "transport=" ( "udp" | "tcp" | "sctp" | "tls" | other-transport) other-transport = token user-param = "user=" ( "phone" | "ip" | other-user) other-user = token method-param = "method=" Method ttl-param = "ttl=" ttl maddr-param = "maddr=" host other-param = pname [ "=" pvalue ] pname = 1*paramchar pvalue = 1*paramchar paramchar = param-unreserved | unreserved | escaped param-unreserved = "[" | "]" | "/" | ":" | " " | "+" | "$" Various Authors [Page 226] Internet Draft SIP January 28, 2002 headers = "?" header *( " " header ) header = hname "=" hvalue hname = 1*( hnv-unreserved | unreserved | escaped ) hvalue = *( hnv-unreserved | unreserved | escaped ) hnv-unreserved = "[" | "]" | "/" | "?" | ":" | "+" | "$" SIP-message = Request | Response Request = Request-Line *( message-header ) CRLF [ message-body ] Request-Line = Method SP Request-URI SP SIP-Version CRLF Request-URI = SIP-URI | absoluteURI absoluteURI = scheme COLON ( hier-part | opaque-part ) hier-part = ( net-path | abs-path ) [ "?" query ] net-path = "//" authority [ abs-path ] abs-path = "/" path-segments opaque-part = uric-no-slash *uric uric = reserved | unreserved | escaped uric-no-slash = unreserved | escaped | ";" | "?" | ":" | "@" | " " | "=" | "+" | "$" | "," path-segments = segment *( "/" segment ) segment = *pchar *( SEMI param ) param = *pchar pchar = unreserved | escaped | ":" | "@" | " " | "=" | "+" | "$" | "," scheme = alpha *( alpha | digit | "+" | "-" | "." ) authority = server | reg-name server = [ [ userinfo "@" ] hostport ] reg-name = 1*( unreserved | escaped | "$" | "," | ";" | ":" | "@" | " " | "=" | "+" ) query = *uric SIP-Version = "SIP/2.0" message-header = Accept | Accept-Encoding | Accept-Language | Alert-Info | Allow | Authentication-Info | Authorization Various Authors [Page 227] Internet Draft SIP January 28, 2002 | Call-ID | Call-Info | Contact | Content-Disposition | Content-Encoding | Content-Language | Content-Length | Content-Type | CSeq | Date | Error-Info | Expires | From | In-Reply-To | Max-Forwards | MIME-Version | Min-Expires | Organization | Priority | Proxy-Authenticate | Proxy-Authorization | Proxy-Require | RAck | Record-Route | Reply-To | Require | Retry-After | Route | RSeq | Server | Subject | Supported | Timestamp | To | Unsupported | User-Agent | Via | Warning | WWW-Authenticate Method = "INVITE" | "ACK" | "OPTIONS" | "BYE" | "CANCEL | "REGISTER" | "PRACK" | extension-method extension-method = token option-tag = token Various Authors [Page 228] Internet Draft SIP January 28, 2002 Response = Status-Line *( message-header ) CRLF [ message-body ] Status-Line = SIP-version SP Status-Code SP Reason-Phrase CRLF Status-Code = Informational | Redirection | Success | Client-Error | Server-Error | Global-Failure | extension-code extension-code = 3DIGIT Reason-Phrase = *(reserved | unreserved | escaped | SP | HT) Informational = "100" ; Trying | "180" ; Ringing | "181" ; Call Is Being Forwarded | "182" ; Queued | "183" ; Session Progress Success = "200" ; OK Redirection = "300" ; Multiple Choices | "301" ; Moved Permanently | "302" ; Moved Temporarily | "305" ; Use Proxy | "380" ; Alternative Service Various Authors [Page 229] Internet Draft SIP January 28, 2002 Client-Error = "400" ; Bad Request | "401" ; Unauthorized | "402" ; Payment Required | "403" ; Forbidden | "404" ; Not Found | "405" ; Method Not Allowed | "406" ; Not Acceptable | "407" ; Proxy Authentication Required | "408" ; Request Timeout | "409" ; Conflict | "410" ; Gone | "413" ; Request Entity Too Large | "414" ; Request-URI Too Large | "415" ; Unsupported Media Type | "416" ; Unsupported URI Scheme | "420" ; Bad Extension | "423" ; Registration Too Brief | "480" ; Temporarily not available | "481" ; Call Leg/Transaction Does Not Exist | "482" ; Loop Detected | "483" ; Too Many Hops | "484" ; Address Incomplete | "485" ; Ambiguous | "486" ; Busy Here | "487" ; Request Terminated | "488" ; Not Acceptable Here | "491" ; Request Pending | "493" ; Undecipherable Server-Error = "500" ; Internal Server Error | "501" ; Not Implemented | "502" ; Bad Gateway | "503" ; Service Unavailable | "504" ; Server Time-out | "505" ; SIP Version not supported Global-Failure = "600" ; Busy Everywhere | "603" ; Decline | "604" ; Does not exist anywhere | "606" ; Not Acceptable Various Authors [Page 230] Internet Draft SIP January 28, 2002 Accept = "Accept" HCOLON ( accept-range *(COMMA accept-range) ) accept-range = media-range [ accept-params ] media-range = ( "*/*" | ( m-type SWS "/" "*" SWS ) | ( m-type SLASH m-subtype ) ) *( SEMI parameter ) accept-params = SEMI "q" EQUAL qvalue *( accept-extension ) accept-extension = SEMI ae-name [ EQUAL ae-value ] ae-name = token ae-value = token | quoted-string Accept-Encoding = "Accept-Encoding" HCOLON ( encoding *(COMMA encoding) ) encoding = codings [ SEMI "q" EQUAL qvalue ] codings = content-coding | "*" content-coding = token qvalue = ( "0" [ "." 0*3DIGIT ] ) | ( "1" [ "." 0*3("0") ] ) Accept-Language = "Accept-Language" HCOLON ( language *(COMMA language) ) language = language-range [ SEMI "q" EQUAL qvalue ] language-range = ( ( 1*8ALPHA *( MINUS 1*8ALPHA ) ) | "*" ) Alert-Info = "Alert-Info" HCOLON alert-param *(COMMA alert-param) alert-param = LAQUOT URI RAQUOT *( SEMI generic-param ) generic-param = token [ EQUAL gen-value ] gen-value = token | host | quoted-string Allow = "Allow" HCOLON Method *(COMMA Method) Authorization = "Authorization" HCOLON credentials credentials = ("Digest" digest-response) | (token gen-resp)) digest-response = dig-resp *(COMMA dig-resp) dig-resp = username | realm | nonce | digest-uri Various Authors [Page 231] Internet Draft SIP January 28, 2002 | dresponse | [ algorithm ] | [cnonce] | [opaque] | [message-qop] | [nonce-count] | [auth-param] username = "username" EQUAL username-value username-value = quoted-string digest-uri = "uri" EQUAL digest-uri-value digest-uri-value = request-uri ; As specified by HTTP/1.1 message-qop = "qop" EQUAL qop-value cnonce = "cnonce" EQUAL cnonce-value cnonce-value = nonce-value nonce-count = "nc" EQUAL nc-value nc-value = 8LHEX dresponse = "response" EQUAL request-digest request-digest = LDQUOT 32LHEX RDQUOT auth-param = auth-param-name EQUAL ( token | quoted-string ) auth-param-name = token gen-resp = *token *((COMMA *token) | (EQUAL (*token | quoted-string)) AuthenticationInfo __ "Authentication-Info" COLON ainfo *(COMMA ainfo) ainfo = nextnonce | [ message-qop ] | [ response-auth ] | [ cnonce ] | [nonce-count] nextnonce "nextnonce" EQUAL nonce-value response-auth = "rspauth" EQUAL response-digest response-digest = LDQUOT *LHEX RDQUOT Call-ID = ( "Call-ID" | "i" ) HCOLON callid callid = word [ "@" word ] Call-Info = "Call-Info" HCOLON info *(COMMA info) info = LAQUOT URI RAQUOT *( SEMI info-param) info-param = "purpose" EQUAL ( "icon" | "info" | "card" | token ) | generic-param Contact = ("Contact" | "m" ) HCOLON Various Authors [Page 232] Internet Draft SIP January 28, 2002 (STAR | contact-param *(COMMA contact-param)) contact-param = name-addr | addr-spec *(SEMI contact-params) name-addr = [ display-name ] LAQUOT addr-spec RAQUOT addr-spec = SIP-URI | URI display-name = *(token LWS)| quoted-string) contact-params = c-p-q | c-p-expires | contact-extension c-p-q = "q" EQUAL qvalue c-p-expires = "expires" EQUAL delta-seconds contact-extension = generic-param qvalue = ( "0" [ "." 0*3DIGIT ] ) | ( "1" [ "." 0*3("0") ] ) delta-seconds = 1*DIGIT Content-Disposition = "Content-Disposition" HCOLON disposition-type *( SEMI disposition-param ) disposition-type = "render" | "session" | "icon" | "alert" | disp-extension-token disposition-param = "handling" EQUAL ( "optional" | "required" | other-handling ) | generic-param other-handling = token disp-extension-token = token Content-Encoding = ( "Content-Encoding" | "e" ) HCOLON content-coding *(COMMA content-coding) Content-Language = "Content-Language" HCOLON language-tag *(COMMA language-tag) language-tag = primary-tag *( MINUS subtag ) primary-tag = 1*8ALPHA Various Authors [Page 233] Internet Draft SIP January 28, 2002 subtag = 1*8ALPHA Content-Length = ( "Content-Length" | "l" ) HCOLON 1*DIGIT Content-Type = ( "Content-Type" | "c" ) HCOLON media-type media-type = m-type SLASH m-subtype *(SEMI m-parameter) m-type = discrete-type | composite-type discrete-type = "text" | "image" | "audio" | "video" | "application" | extension-token composite-type "message" | "multipart" | extension-token extension-token = ietf-token | x-token ietf-token = token x-token = ("X" | "x") "-" token m-subtype = extension-token | iana-token iana-token = token m-parameter = m-attribute EQUAL m-value m-attribute = token m-value = token | quoted-string CSeq = "CSeq" HCOLON 1*DIGIT LWS Method Date = "Date" HCOLON SIP-date SIP-date = rfc1123-date rfc1123-date = wkday COMMA date1 SP time SP "GMT" date1 = 2DIGIT SP month SP 4DIGIT ; day month year (e.g., 02 Jun 1982) time = 2DIGIT ":" 2DIGIT ":" 2DIGIT ; 00:00:00 - 23:59:59 wkday = "Mon" | "Tue" | "Wed" | "Thu" | "Fri" | "Sat" | "Sun" month = "Jan" | "Feb" | "Mar" | "Apr" | "May" | "Jun" | "Jul" | "Aug" | "Sep" | "Oct" | "Nov" | "Dec" Various Authors [Page 234] Internet Draft SIP January 28, 2002 Error-Info = "Error-Info" HCOLON error-uri *(COMMA error-uri) error-uri = LAQUOT URI RAQUOT *( SEMI generic-param ) Expires = "Expires" HCOLON delta-seconds >From = ( "From" | "f" ) HCOLON from-spec from-spec = ( name-addr | addr-spec ) *( SEMI from-param ) from-param = tag-param | generic-param tag-param = "tag" EQUAL token In-Reply-To = "In-Reply-To" HCOLON called *(COMMA called) Max-Forwards = "Max-Forwards" HCOLON 1*DIGIT MIME-Version = "MIME-Version" HCOLON 1*DIGIT "." 1*DIGIT Min-Expires = "Min-Expires" HCOLON delta-seconds Organization = "Organization" HCOLON TEXT-UTF8-TRIM Priority = "Priority" HCOLON priority-value priority-value = "emergency" | "urgent" | "normal" | "non-urgent" | other-priority other-priority = token Various Authors [Page 235] Internet Draft SIP January 28, 2002 Proxy-Authenticate = "Proxy-Authenticate" HCOLON challenge *(COMMA challenge) challenge = "Digest" digest-challenge digest-challenge = digest-chlng *(COMMA digest-chlng) digest-chlng = realm | [ domain ] | nonce | [ opaque ] | [ stale ] | [ algorithm ] | [ qop-options ] | [auth-param] realm = "realm" EQUALS realm-value realm-value = quoted-string domain = "domain" EQUAL LDQUOT URI ( 1*SP URI ) RDQUOT URI = absoluteURI | abs_path nonce = "nonce" EQUAL nonce-value nonce-value = quoted-string opaque = "opaque" EQUAL quoted-string stale = "stale" EQUAL ( "true" | "false" ) algorithm = "algorithm" EQUAL ( "MD5" | "MD5-sess" | token ) qop-options = "qop" EQUAL LDQUOT qop-value *(COMMA qop-value) RDQUOT qop-value = "auth" | "auth-int" | token Proxy-Authorization = "Proxy-Authorization" HCOLON credentials Proxy-Require = "Proxy-Require" HCOLON option-tag *(COMMA option-tag) RAck = "RAck" HCOLON response-num LWS CSeq-num LWS Method response-num = 1*DIGIT CSeq-num = 1*DIGIT response-num = 1*DIGIT Record-Route = "Record-Route" HCOLON rec-route *(COMMA rec-route) rec-route = name-addr *( SEMI rr-param ) rr-param = generic-param Various Authors [Page 236] Internet Draft SIP January 28, 2002 Reply-To = ( "Reply-To" | "f" ) HCOLON rplyto-spec rplyto-spec = ( name-addr | addr-spec ) *( SEMI rplyto-param ) rplyto-param = generic-param Require = "Require" HCOLON option-tag *(COMMA option-tag) Retry-After = "Retry-After" HCOLON delta-seconds [ comment ] *( SEMI retry-param ) retry-param = "duration" EQUAL delta-seconds | generic-param Route = "Route" HCOLON route=param *(COMMA route-param) route-param = name-addr *( SEMI rr-param ) RSeq = "RSeq" HCOLON response-num Server = "Server" HCOLON 1*( product | comment ) product = token [SLASH product-version] product-version = token Subject = ( "Subject" | "s" ) HCOLON TEXT-UTF8-TRIM Supported = ( "Supported" | "k" ) HCOLON (option-tag *(COMMA option-tag) Timestamp = "Timestamp" HCOLON 1*(DIGIT) [ "." *(DIGIT) ] [ delay ] delay = *(DIGIT) [ "." *(DIGIT) ] Various Authors [Page 237] Internet Draft SIP January 28, 2002 To = ( "To" | "t" ) HCOLON ( name-addr | addr-spec ) *( SEMI to-param ) to-param = tag-param | generic-param Unsupported = "Unsupported" HCOLON option-tag *(COMMA option-tag) User-Agent = "User-Agent" HCOLON 1*( product | comment ) Via = ( "Via" | "v" ) HCOLON via-parm *(COMMA via-parm) via-parm = sent-protocol sent-by *( SEMI via-params ) via-params = via-ttl | via-maddr | via-received | via-branch | via-extension via-ttl = "ttl" EQUAL ttl via-maddr = "maddr" EQUAL host via-received = "received" EQUAL (IPv4address | IPv6address) via-branch = "branch" EQUAL token via-extension = generic-param sent-protocol = protocol-name SLASH protocol-version SLASH transport protocol-name = "SIP" | token protocol-version = token transport = "UDP" | "TCP" | "TLS" | "SCTP" | other-transport sent-by = host [ COLON port ] ttl = 1*3DIGIT ; 0 to 255 Warning = "Warning" HCOLON warning-value *(COMMA warning-value) warning-value = warn-code SP warn-agent SP warn-text warn-code = 3DIGIT warn-agent = ( host [ COLON port ] ) | pseudonym ; the name or pseudonym of the server adding ; the Warning header, for use in debugging warn-text = quoted-string pseudonym = token Various Authors [Page 238] Internet Draft SIP January 28, 2002 WWW-Authenticate = "WWW-Authenticate" HCOLON challenge message-body = *OCTET 28 IANA Considerations All new or experimental method names, header field names, and status codes used in SIP applications SHOULD be registered with IANA in order to prevent potential naming conflicts. It is RECOMMENDED that new "option- tag"s and "warn-code"s also be registered. Before IANA registration, new protcol elements SHOULD be described in an Internet-Draft or, preferably, an RFC. For Internet-Drafts, IANA is requested to make the draft available as part of the registration database. By the time an RFC is published, colliding names may have already been implemented. When a registration for either a new header field, new method, or new status code is created based on an Internet-Draft, and that Internet-Draft becomes an RFC, the person that performed the registration MUST notify IANA to change the registration to point to the RFC instead of the Internet-Draft. Registrations should be sent to iana@iana.org 28.1 Option Tags Option tags are used in header fields such as Require, Supported, Proxy-Require, and Unsupported in support of SIP compatibility mechanisms for extensions ( Section 23.2). The option tag itself is a string that is associated with a particular SIP option (that is, an extension). It identifies the option to SIP endpoints. When registering a new SIP option with IANA, the following information MUST be provided: o Name and description of option. The name MAY be of any length, but SHOULD be no more than twenty characters long. The name MUST consist of alphanum (Section 27) characters only. o A listing of any new SIP header fields, header parameter fields, or parameter values defined by this option. A SIP Various Authors [Page 239] Internet Draft SIP January 28, 2002 option MUST NOT redefine header fields or parameters defined in either RFC 2543, any standards-track extensions to RFC 2543, or other extensions registered through IANA. o Indication of who has change control over the option (for example, IETF, ISO, ITU-T, other international standardization bodies, a consortium, or a particular company or group of companies). o A reference to a further description if available, for example (in order of preference) an RFC, a published paper, a patent filing, a technical report, documented source code, or a computer manual. o Contact information (postal and email address). This procedure has been borrowed from RTSP [3] and the RTP AVP [41]. 28.1.1 Registration of 100rel This specification registers a single option tag, "100rel". The required information is: Name: "100rel" Description: This option tag is for reliability of provisional responses. When present in a Supported header, it indicates that the UA can send or receive reliable provisional responses. When present in a Require header in a request, it indicates that the UAS MUST send all provisional responses reliably. When present in a Require header in a reliable provisional response, it indicates that the response is to be sent reliably. New Headers: The RSeq and RAck header fieds are defined by this optio. Change Control: IETF. Reference: RFCXXXX [Note to IANA: Fill in with the RFC number of this specification. Contact Information: Jonathan Rosenberg, jdrosen@jdrosen.net. 72 Eagle Rock Avenue, First Floor, East Hanover, NJ, 07936, USA. Various Authors [Page 240] Internet Draft SIP January 28, 2002 28.2 Warn-Codes Warning codes provide information supplemental to the status code in SIP response messages when the failure of the transaction results from a Session Description Protocol (SDP, [5]). New "warn-code" values can be registered with IANA as they arise. The "warn-code" consists of three digits. A first digit of "3" indicates warnings specific to SIP. Warnings 300 through 329 are reserved for indicating problems with keywords in the session description, 330 through 339 are warnings related to basic network services requested in the session description, 370 through 379 are warnings related to quantitative QoS parameters requested in the session description, and 390 through 399 are miscellaneous warnings that do not fall into one of the above categories. 1xx and 2xx have been taken by HTTP/1.1. 28.3 Header Field Names Header field names do not require working group or working group chair review prior to IANA registration, but SHOULD be documented in an RFC or Internet-Draft before IANA is consulted. The following information needs to be provided to IANA in order to register a new header field name: o The name and email address of the individual performing the registration; o the name of the header field being registered; o a compact form version for that header field, if one is defined; o the name of the draft or RFC where the header field is defined; o a copy of the draft or RFC where the header field is defined. Header fields SHOULD NOT use the X prefix notation and MUST NOT duplicate the names of header fields used by SMTP or HTTP unless the syntax is a compatible superset and the semantics are similar. Some common and widely used header fields MAY be assigned one-letter compact forms (Section 7.3.3). Compact forms can only be assigned Various Authors [Page 241] Internet Draft SIP January 28, 2002 after SIP working group review. In the absence of this working group, a designated expert reviews the request. 28.4 Method and Response Codes Because the status code space is limited, they do require working group or working group chair review, and MUST be documented in an RFC or Internet draft. The same procedures apply to new method names. The following information needs to be provided to IANA in order to register a new response code or method: o The name and email address of the individual performing the registration; o the number of the response code or name of the method being registered; o the default reason phrase for that status code, if applicable; o the name of the draft or RFC where the method or status code is defined; o a copy of the draft or RFC where the method or status code is defined. 29 Changes Made in Version 00 o Indicated that UAC should send both CANCEL and BYE after a retransmission fails. o Added semicolon and question mark to the list of unreserved characters for the user part of SIP URLs to handle tel: URLs properly. o Uniform handling of if hop count Max-Forwards: return 483. Note that this differs from HTTP/1.1 behavior, where only OPTIONS and TRACE allow this header, but respond as the final recipient when the value reaches zero. o Clarified that a forking proxy sends ACKs only for INVITE requests. o Clarified wording of DNS caching. Added paragraph on "negative caching", i.e., what to do if one of the hosts failed. It is probably not a good idea to simply drop this host from the list if the DNS ttl value is more than a few minutes, since that would mean that load balancing may not work for quite a Various Authors [Page 242] Internet Draft SIP January 28, 2002 while after a server is brought back on line. This will be true in particular if a server group receives a large number of requests from a small number of upstream servers, as is likely to be the case for calls between major consumer ISPs. However, without getting into arbitrary and complicated retry rules, it seems hard to specify any general algorithm. Might it be worthwhile to simply limit the "black list" interval to a few minutes? o Added optional Call-Info and Alert-Info header fields that describe the caller and information to be used in alerting. (Currently, avoided use of "purpose" qualification since it is not yet clear whether rendering content without understanding its meaning is always appropriate. For example, if a UAS does not understand that this header is to replace ringing, it would mix both local ring tone and the indicated sound URL.) TBD! o SDP "s=" lines can't be empty, unfortunately. o Noted that maddr could also contain a unicast address, but SHOULD contain the multicast address if the request is sent via multicast (Section 24.44. o Clarified that responses are sent to port in Via sent-by value. o Added "other-*" to the user URL parameter and the Hide and Content-Disposition headers. o Clarified generation of timeout (408) responses in forking proxies and mention the Expires header. o Clarified that CANCEL and INVITE are separate transactions (Fig. 7). Thus, the INVITE request generates a 487 (Request Terminated) if a CANCEL or BYE arrives. o Clarified that Record-Route SHOULD be inserted in every request, but that the route, once established, persists. This provides robustness if the called UAS crashes. o Emphasized that proxy, redirect, registrar and location servers are logical, not physical entities and that UAC and UAS roles are defined on a request-by-request basis. (Section 6) o In Section 24.44, noted that the maddr and received parameters also need to be encrypted when doing Via hiding. Various Authors [Page 243] Internet Draft SIP January 28, 2002 o Simplified Fig. 7 to only show INVITE transaction. o Added definition of the use of Contact (Section 24.10) for OPTIONS. o Added HTTP/RFC 822 headers Content-Language and MIME-Version. o Added note in minimal section indicating that UAs need to support UDP. o Added explanation explaining what a UA should do when receiving an initial INVITE with a tag. o Clarified UA and proxy behavior for 302 responses. o Added details on what a UAS should do when receiving a tagged INVITE request for an unknown call leg. This could occur if the UAS had crashed and the UAC sends a re-INVITE or if the BYE got lost and the UAC still believes to be in the call. o Added definition of Contact in 4xx, 5xx and 6xx to "redirect" to more error details. o Added note to forking proxy description to gather *- Authenticate from responses. This allows several branches to be authenticated simultaneously. o Changed URI syntax to use URL escaping instead of quotation marks. o Changed SIP URL definition to reference RFC 2806 for telephone-subscriber part. o Clarified that the To URI should basically be ignored by the receiving UAS except for matching requests to call legs. In particular, To headers with a scheme or name unknown to the callee should be accepted. o Clarified that maddr is to be added by any client, either proxy or UAC. o Added response code 488 to indicate that there was no common media at the particular destination. (606 indicates such failure globally.) o In Section 24.19, noted that registration updates can shorten the validity period. Various Authors [Page 244] Internet Draft SIP January 28, 2002 o Added note to enclose the URI for digest in quotation marks. The BNF in RFC 2617 is in error. o Clarified that registrars use Authorization and WWW- Authenticate, not proxy authentication. o Added note in Section 24.10 that "headers" are copied from Contact into the new request. o Changed URL syntax so that port specifications have to have at least one digit, in line with other URL formats such as "http". Previously, an empty port number was permissible. o In SDP section, added a section on how to add and delete streams in re-INVITEs. o IETF-blessed extensions now have short names, without org.ietf. prefix. o Cseq is unique within a call leg, not just within a call (Section 24.16). o Added IPv6 literal addresses to the SIP URL definition, according to RFC 2732 [42]. Modified the IPv4 address to limit segments to at most three digits. o Modified registration procedure so that it explicitly references the URL comparison. Updates with shorter expiration time are now allowed. o For send-only media, SDP still must indicate the address and port, since these are needed as destinations for RTCP messages. o Changed references regarding DNS SRV records from RFC 2052 to RFC 2782, which is now a Proposed Standard. Integrated SRV into the search procedure and removed the SRV appendix. The only visible change is that protocol and service names are now prefixed by an underscore. Added wording that incorporates the precedence of maddr. o Allow parameters in Record-Route and Route headers. o In Table 1, list udp as the default value for the transport parameter in SIP URI. o Removed sentence that From can be encrypted. It cannot, since the header is needed for call-leg identification. Various Authors [Page 245] Internet Draft SIP January 28, 2002 o Added note that a UAC only copies a To tag into subsequent transactions if it arrives in a 200 OK to an INVITE. This avoids the problem that occurs when requests get resubmitted after receiving, say, a 407 (or possibly 500, 503, 504, 305, 400, 411, 413, maybe even 408). Under the old rules, these requests would have a tag, which would force the called UAS to reject the request, since it doesn't have an entry for this tag. o Loop detection has been modified to take the request-URI into account. This allows the same request to visit the server twice, but with different request URIs ("spiral"). o Elaborated on URL comparison and comparison of From/To fields. o Added np-queried user parameter. o Changed tag syntax from UUID to token, since there's no reason to restrict it to hex. o Added Content-Disposition header based on earlier discussions about labeling what to do with a message body (part). o Clarification: proxies must insert To tags for locally generated responses. o Clarification: multicast may be used for subsequent registrations. o Feature: Added Supported header. Needed if client wants to indicate things the server can usefully return in the response. o Bug: The From, To, and Via headers were missing extension parameters. The Encryption and Response-Key header fields now "officially" allow parameters consisting only of a token, rather than just "token = value". o Bug: Allow was listed as optional in 405 responses in Table 2. It is mandatory. o Added: "A BYE request from either called or calling party terminates any pending INVITE, but the INVITE request transaction MUST be completed with a final response." o Clarified: "If an INVITE request for an existing session fails, the session description agreed upon in the last successful INVITE transaction remains in force." Various Authors [Page 246] Internet Draft SIP January 28, 2002 o Clarified what happens if two INVITE requests meet each other on the wire, either traveling the same or in opposite directions: A UAC MUST NOT issue another INVITE request for the same call leg before the previous transaction has completed. A UAS that receives an INVITE before it sent the final response to an INVITE with a lower CSeq number MUST return a 400 (Bad Request) response and MUST include a Retry-After header field with a randomly chosen value of between 0 and 10 seconds. A UA that receives an INVITE while it has an INVITE transaction pending, returns a 500 (Internal Server Error) and also includes a Retry-After header field. o Expires header clarified: limits only duration of INVITE transaction, not the actual session. SDP does the latter. o The In-Reply-To header was added. o There were two incompatible BNFs for WWW-Authenticate. One defined for PGP, and the other borrowed from HTTP. For basic or digest: WWW-Authenticate: basic realm="Wallyworld" and for pgp: WWW-Authenticate: pgp; realm="Wallyworld" The latter is incorrect and the semicolon has been removed. o Added rules for Route construction from called to calling UA. o We now allow Accept and Accept-Encoding in BYE and CANCEL requests. There is no particular reason not to allow them, as both requests could theoretically return responses, particularly when interworking with other signaling systems. o PGP "pgp-pubalgorithm" allows server to request the desired public-key algorithm. Various Authors [Page 247] Internet Draft SIP January 28, 2002 o ABNF rules now describe tokens explicitly rather than by subtraction; explicit character enumeration for CTL, etc. o Registrars should be careful to check the Date header as the expiration time may well be in the past, as seen by the client. o Content-Length is mandatory; Table 2 erroneously marked it as optional. o User-Agent was classified in a syntax definition as a request header rather than a general header. o Clarified ordering of items to be signed and include realm in list. o Allow Record-Route in 401 and 484 responses. o Hop-by-hop headers need to precede end-to-end headers only if authentication is used. o 1xx message bodies MAY now contain session descriptions. o Changed references to HTTP/1.1 and authentication to point to the latest RFCs. o Added 487 (Request terminated) status response. It is issued if the original request was terminated via CANCEL or BYE. o The spec was not clear on the identification of a call leg. Section 1.3 says it's the combination of To, From, and Call- ID. However, requests from the callee to the caller have the To and From reversed, so this definition is not quite accurate. Additionally, the "tag" field should be included in the definition of call leg. The spec now says that a call leg is defined as the combination of local-address, remote- address, and call-id, where these addresses include tags. Text was added to Section 6.21 to emphasize that the From and To headers designate the originator of the request, not that of the call leg. o All URI parameters, except method, are allowed in a Request- URI. Consequently, also updated the description of which parameters are copied from 3xx responses in Sec. 24.10. o The use of CRLF, CR,or LF to terminate lines was confusing. Basically, each header line can be terminated by a CR, LF, or Various Authors [Page 248] Internet Draft SIP January 28, 2002 CRLF. Furthermore, the end of the headers is signified by a "double return". Simplified to require sending of CRLF, but require senders to receive CR and LF as well and only allow CR CR, LF LF in addition to double CRLF as a header-body separator. o Round brackets in Contact header were part of the HTTP legacy, and very hard to implement. They are also not that useful and were removed. o The spec said that a proxy is a back-to-back UAS/UAC. This is almost, but not quite, true. For example, a UAS should insert a tag into a provisional response, but a proxy should not. This was clarified. o Section 6.13 in the RFC begins mid-paragraph after the BNF. The following text was misplaced in the conversion to ASCII: Even if the "display-name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, semicolon or question mark. 30 Changes Made in Version 01 o Uniform syntax specification for semicolon parameters: Foo = "Foo" ":" something *( ";" foo-param ) foo-param = "bar" "=" token | generic-param o Removed np-queried user parameter since this is now part of a tel URL extension parameter. o In SDP section, noted that if the capabilities intersection is empty, a dummy format list still has to be returned due to SDP syntax constraints. Previously, the text had required that no formats be listed. (Brian Rosen) o Reorganized tables 2 and 3 to show proxy interaction with headers rather than "end-to-end" or "hop-by-hop". 31 Changes Made in Version 02 o Added "or UAS" in description of received headers in Section 24.44. This makes the response algorithm work even if the last IP address in the Via is incorrect. Various Authors [Page 249] Internet Draft SIP January 28, 2002 o Tentatively removed restriction that CANCEL requests cannot have Route headers. (Billy Biggs) o Tentatively added Also header for BYE requests, as it is widely implemented and a simple means to implement unsupervised call transfer. Subject to removal if there is protest. (Billy Biggs) o If a proxy sends a request by UDP (TCP), the spec did not disallow placing TCP (UDP) in the transport parameter of the Via field, which it should. Added a note that the transport protocol actually used is included. o No default value for the q parameter in Contact is defined. This is not strictly needed, but is useful for consistent behaviors at recursive proxies and at UAC's. Now 0.5. o Clarified that To and From tag values should be different to simplify request matching when calling oneself. o Removed ability to carry multiple requests in a single UDP packet (Section 24.14). o Added note that Allow MAY be included in requests, to indicate requestor capabilities for the same call ID. o Added note to Section 24.17 indicating that registrars MUST include the Date header to accomodate UAs that do not have a notion of absolute time. o Added note emphasizing that non-SIP URIs are permissible in REGISTER. o Rewrote the server lookup section to be more precise and more like pseudo-code, with nesting instead of "gotos". o Removed note Note that the two URLs example.com and example.com:5060, while considered equal, may not lead to the same server, as the former causes a DNS SRV lookup, while the latter only uses the A record. since that is no longer the case. o Emphasized that proxies have to forward requests with unknown methods. o Aligned definition of call leg with URI comparison rules. Various Authors [Page 250] Internet Draft SIP January 28, 2002 o Required that second branch parameter be globally unique, so that a proxy can distinguish different branches in spiral scenarios similar to the following, with record-routing in place: B ---> P1 -------> P2 ------------> P1 ----------------> A BYE B B/1 P1/2,B/1 P2/3,P1/2,B/1 P1/4,P2/3,P1/2,B/1 Here, A/1 denotes the Via entry with host A and branch parameter 1. Also, this requires updating the definition of isomorphic requests, since the Request-URI is the same for all BYE that are record-routed. o Removed Via hiding from spec, for the following reasons: - complexity, particularly hidden "gotchas" that surface at various points (as in this instance); - interference with loop detection and debugging; - Unlike HTTP, where via-hiding makes sense since all data is contained in the request or response, Via-hiding in SIP by itself does nothing to hide the caller or callee, as address information is revealed in a number of places: - Contact; - Route/Record-Route; - SDP, including the o= and c= lines; - possibly accidental leakage in User-Agent header and Call-ID headers. - Unless this is implemented everywhere, the feature is not likely to be very useful, without the sender having any recourse such as "don't route this request unless you can hide". It appears that almost all existing proxies simply ignore the Hide header. o Added Error-Info header field. 32 Changes Made in Version 03 o Description of Route and Record-Route moved to separate section, which is new. All UAs must now support this mechanism. Various Authors [Page 251] Internet Draft SIP January 28, 2002 o Removed status code 411, since it cannot occur (Jonathan Rosenberg, James Jack). o Rewrote Record-Route section to reflect new mechanism. In particular, requests from callee to caller now use the same path as in the opposite direction, without substituting the From header field values. The maddr parameter is now optional. o Disallowed SIP URLs that only have a password, without a user name. The prototype from RFC 1738 also doesn't allow this. o Allow registrar to set the expiration time. o CSeq (Section 24.16) is counted within a call leg, not a call. o Removed wording that connection closing is equivalent to CANCEL or 500. This does not work for connections that are used for multiple transactions and has other problems. o Cleaned up CSeq section. Removed text about inserting CSeq method when it is absent. Clarified that CSeq increments for all requests, not just INVITE. Clarified that all out of order requests, not just out of order INVITE, are rejected with a 400 class response. Clarified the meaning of "initial" sequence number. Clarified that after a request forks, each 200 OK is a separate call leg, and thus, separate CSeq space. Clarified that CSeq numbers are independent for each direction of a call leg. o Massive reorganization and cleanup of the SDP section. Introduced the concept of the offer-answer model. Clarified that set of codecs in m line are usable all at the same time. Inserted size restriction on representation of values in o line. Explicitly describe forked media. New media lines for adding streams appear at the bottom of the SDP (used to say append). o Removed Also. o Added text to Require and Proxy-Require sections, making it a SHOULD to retry the request without the unsupported extension. o Added text to section on 415, saying that UAC SHOULD retry the request without the unsupported body. o Added text to section on CANCEL and ACK, clarifying much of the behavior. Various Authors [Page 252] Internet Draft SIP January 28, 2002 o Modified Content-Type to indicate that it can be present even if the body is empty. o From tags mandatory o Old text said that if you hang up before sending an ACK, you need not send the ACK. That is wrong. Text fixed so that an ACK is always sent. o Old text said that if you never got a response to an INVITE, the UAC should send both an INVITE and CANCEL. This doesn't make sense. Rahter, it should do nothing and consider the call terminated. o Added text that says pending requests are responded to with a 487 if a BYE is received. o Updated section 2.2, so that its clear that Contact is not used with BYE. o Clarified Via processing rules. Added text on handling loops when proxies route on headers besides the request URI. Added text on handling case when sent-by contains a domain name. Added text to 6.47 on opening TCP connections to send responses upstream. o Clarified that a 1xx with an unknown xx is not the same as the 100 response. o Removed usage of Retry-After in REGISTER. o Clarified usage of persistent connections. o Clarified that servers supporting HTTP basic or digest in rfc2617 MUST be backwards compatible with RFC 2069. o Clarified that ACK contains the same branch ID as the request its acknowledging. o Added definitions for spiral, B2BUA. o Rephrased definitions for UAC, UAS, Call, call-leg, caller, callee, making them more concrete. o URL comparison ignores parameters not present in both URLs only for unknown parameters. o Clarified that * in Contact is used only in REGISTER with Various Authors [Page 253] Internet Draft SIP January 28, 2002 Expires header zero. Mentioned * case in section on Contact syntax. o Removed text that says a UA can insert a Contact in 2xx that indicates the address of a proxy. Not likely to work in general. o Removed SDP text about aligning media streams within a media type to handle certain crash and restart cases. o Receiving a 481 to a mid-call request terminates that call leg. Agreed upon at IETF 49. o Introduced definition of regular transaction - non-INVITE excepting ACK and CANCEL. o Clarified rules for overlapping transactions. o Forking proxies MUST be stateful (used to say SHOULD). Proxies that send requests on multicast MUST be stateful (used to say nothing) o Text added recommending that registrars authorize that entity in From field can register address-of-record in the To field. o Forwarding of non-100 provisionals upstream in a proxy changed from SHOULD to MUST. o Removed PGP. 33 Changes Made in Version 04 o Removed Unsupported as a request header from Table 3. o Clarified SDP procedures for changing IP address and port. Specifically, spelled out the duration for which a UA needs to received media on the old port and address. o Added text in the SDP session which recommends that the answerer use the same ordering of codecs as used on the offer, in order to help ensure symmetric codec operation under normal conditions. o Fixed bug in the example in the SDP section, where the new media line was listed at the top. Should have been the bottom. o Authorization credentials are cached based on the URL of the To header, not the entire To header as 10.48 implied. Various Authors [Page 254] Internet Draft SIP January 28, 2002 o Section 10.31, on Proxy-Authenticate, indicated that a server responds with a 401 if the client guessed wrong. This is incorrect. It should be 407. o Section 10.14, removed motivational text about Contact allowing an INVITE to be routed directly between end systems, since its confusing. Some have interpreted to mean that Record-Route is ignored when Contact is present. o Added reference to SCTP RFC. o Updated 2.2 to allow non-SIP URLs in OPTIONS and 2xx to OPTIONS. o Fixed example in 20.5. Added ACK for 487, and added To tag to 487 response. o Clarified further URL comparisons. Its only URL parameters without defaults that are ignored if not present in both URLs. o Section 1.5.2, UDP mandatory for all. TCP is a SHOULD for UA, MUST for proxy, registrar, redirect servers. o Brought syntax for Contact, Via, and the SIP URL into alignment between the text and postscript versions. o Updated the text in section 6 which said that the ordering of header fields follows HTTP, with the exception of Via, where order matters. However, the HTTP spec says that order matters, so this sentence is redundant and confusing. The sentence was removed. o Added e lines to SDP examples in the Examples section. o Rewrote Allow discussion, more formally defining its semantics and usage cases. o Updated text on 604 status, to indicate that its based on the Request-URI, not the To. o Added response registrations to IANA considerations. Provided more details on registration process. o Clarified that only a UAS rejects a request because the To tag doesn't match a local value. o Clarified that stateless proxies need to route based on static criteria only. Various Authors [Page 255] Internet Draft SIP January 28, 2002 o Proxy and UAC CANCEL generation upon 2xx, 6xx if it forked is now a SHOULD; used to be a MAY. o Added text saying that a UAS SHOULD send a BYE if it never gets an ACK for a 2xx establishing a call leg. o Added text saying that a UAS SHOULD send a re-INVITE if it never gets an ACK for a 2xx to a re-INVITE. o Added text on 503 processing, indicating that a client should try a different server when receiving a 503, and that a proxy shouldn't forward a 503 upstream unless it can't service any other requests. o Removed motivational text in Section 10.43 on Via headers since its not consistent with the text before it. o Changed IPSec reference to RFC 2401, from RFC 1825. o Updated retransmission defininition in 17.3.4 to be consistent with the rest of the spec. o Softened the language for insertion of the transport param in the record-route. Specifically, it can be inserted in private networks where it is known apriori that the specific transport is supported. o Updated definition of B2BUA. o Added text to section on 420 processing, which mandates that the client retry the request without extensions listed in the Unsupported header in the response. o Allow Authentication-Info header to be used for HTTP digest. 34 Changes Made in Version 05 o Updated Table 2 to reflect that Error-Info is a response header in 3xx-6xx responses (it was previously listed as a request header). o Removed WWW-Authenticate as a request header from Table 3. Authentication of responses is now done according to RFC 2617. o Updated the Accept, Accept-Encoding and Accept-Language sections. More details on precise semantics for the various requests and responses is now provided. Presence of these headers is now a SHOULD for INVITE and 2xx to INVITE when a Various Authors [Page 256] Internet Draft SIP January 28, 2002 non-default value is present. Extra emphasis is placed on including the Accept-Language in INVITE and 2xx in order to support internationalization. Usage of these three headers in CANCEL has been removed since it makes no sense. o Generalized local outbound processing rules in Section 16.4.1 to cover the case where the UAS is using a local outbound proxy which was not in the initial call setup path. o Updated record-routing section, so that a proxy can insert a transport param if it knows that the proxy on one side supports the specific transport (the previous text required the proxy to know whether the proxies on both sides supported the specific transport). o Added Authentication-Info to Section 10. o Clarified the meaning of Table 2 for responses. o Updated Table 1 to reflect that maddr is no longer mandatory in Record-Route. o Updated Table 3 so that header fields in responses to ACK are never listed as optional, mandatory, etc. - only not applicable. This is because responses to ACK are not allowed. Also improved wording in Section 5.1.1 to clarify that there MUST NOT be responses to ACK. o Updated SRV procedures. Old text said to treat a failure to contact a server as a 4xx, which would stop the SRV processing. But, this is not so. Sentence was stricken. o Updated 12.1 to clarify that 2xx INVITE responses MUST contain session descriptions. o Changed User-Agent to a request header in Table 3. o Updated SDP section, so that a UA cannot change the SDP when it gets a re-INVITE with no SDP. o Clarified Appendix B that a unicast offer MUST have a unicast response. o Clarified that any request can be record-routed, but it may not be used by the UA, depending on the method. o non-2xx responses to INVITE no longer retransmitted over TCP. Various Authors [Page 257] Internet Draft SIP January 28, 2002 o Removed lower bound on T1 and T2 in private networks, which can use lower values. Furthermore, T1 can be smaller on the public Internet if proper RTT estimation is used. o UAS Cannot send a BYE for a call leg until it receives ACK, in order to eliminate a race condition between BYE and 200 OK. o Support of CR or LF alone as line terminators, as opposed to CRLF, is no longer required. o Client behavior on receipt of a 3xx to re-INVITE is now specified, and it is no longer forbidden to generate a 3xx. This is needed to maintain the idempotency of INVITE, as a proxy might redirect without knowing its a 3xx. o CANCEL cannot be sent before a 1xx is received, in order to eliminate race condition between request and CANCEL. o Termination of the client and server transactions is now based entirely on timeouts, rather than retransmission counters, in order to unify TCP and UDP behavior. Timeout values scale as a function of the RTT estimate, defined as T1. For reliable transports, many of these timers are now set to zero. Many timeouts differ than in bis-04. o Added a working RTT estimation algorithm using the Timestamp header, and specified it to be compliant to RFC 2988. o UAS accepting requests with unknown schemes in the URI in the To field is now a RECOMMENDED instead of SHOULD. This reflects the fact that processing a request when the To field doesn't match is a matter of policy. o Bodies are now allowed in any request and response, including CANCEL, although there may not be any semantics associated with that. o Supporting of INVITE without SDP is now a MUST (no strength was previously specified). o Registration procedures for visiting, which had a few sentences in bis-04, have been removed. Roaming is a complex issue, and should be treated elsewhere. o Bis-04 mandated that a 2xx response to REGISTER contain expires Contact parameters indicating the expiration time of a contact. This behavior has now been made consistent with requests, so that the expiration time of a contact is the same Various Authors [Page 258] Internet Draft SIP January 28, 2002 in either case: the expires param is used first if present, then the Expires header if present, else one hour for SIP URLs. o Action parameter in contact registrations is deprecated. o 2xx to REGISTER MUST contain current contacts. This was just a SHOULD in bis-04. o Multicast operation radically changed. Now, the treatment is no different than unicast. That is, only the first non-1xx response to a multicast request will be used. This is a natural consequence of the layering now applied to the protocol. This still enables anycast types of functions, mirroring the real usage of registrar discovery. o To completely separate transport rules from transaction rules, the rule in bis-04 that said a UAC SHOULD keep a connection opened until a response is received, has been turned into a timer recommendation. Specifically, the spec now says that it is RECOMMENDED that connections be kept opened for a minimum interval of sufficient duration to guarantee, with high probability, that responses are sent over the same connections as a request. o Re-use of existing connections for new requests to the same address and port is now RECOMMENDED, it was only a MAY in bis-04. o Modification of headers below the Authorization header by proxies is no longer disallowed, since the only mechanism that used Authorization in that way, PGP, has been deprecated previously. o Authentication of registrations now RECOMMENDED; no strength was defined previously. o Registering of new headers with IANA is now SHOULD; no strength was defined previously. o Proxy aggregation of challenges now a SHOULD; no strength was defined previously. o Server support of basic authentication downgraded from SHOULD to MAY. o UAC resubmitting requests with credentials after a challenge upgraded from MAY to SHOULD. Various Authors [Page 259] Internet Draft SIP January 28, 2002 o TLS is now RECOMMENDED as the transport layer security for SIP signaling. o UA recursion on a redirect is now SHOULD; no strength was assigned previously. o UA reuse of headers in a recursed request is now SHOULD; no strength was assigned previously. o Security considerations added for Call-Info and Alert-Info. o Proxies no longer forward a 6xx immediately on receiving it. Instead, they CANCEL pending branches immediately. This avoids a potential race condition that would result in a UAC getting a 6xx followed by a 2xx. In all cases except this race condition, the result will be the same - the 6xx is forwarded upstream. o The term call-leg has been eliminated from the spec; a more generic term, dialog, is used in its place. o For SRV processing, subsequent requests with the same Call-ID (as opposed to the same transaction in bis-04) are sent to the same server. o SRV processing generalized to deal with the fact that the default port is transport dependent. o Per IESG request, draft-ietf-sip-serverfeatures has been integrated into bis. o Per IESG request, draft-ietf-sip-100rel will be integrated into bis. This is marked with a placeholder in this draft. o The BNF has been converted from implicit LWS to explicit LWS. o Caching of responses in a proxy to avoid redoing location server lookups used to be a SHOULD. Caching behavior for responses is now fully encapsulated in the transaction processing. o Proxy usage of SRV in processing Route headers upgraded from SHOULD to MUST. 35 Changes Made in Version 06 o Made TCP mandatory for user agents. Various Authors [Page 260] Internet Draft SIP January 28, 2002 o The two states of a dialog are now called early and confirmed. o CANCEL requests now carry Route header fields. o Changes section in -05 forgot to mention the removal of the Encryption and Response-Key headers. These were removed since the only mechanism that used them, PGP, had already been deprecated. As such, they were effectively "garbage collected". o Updated error in transaction definition. ACK-2xx is a separate transaction, ACK for non-2xx is part of the same transaction. o Changed Contact-Length typo to Content-Length in various sections, including throughout the examples section. o Changed Table 3 entry for Record-Route and Route for REGISTER from "o" for optional to "-" for Not Allowed. o Changed Table 3 entry for Route for ACK, BYE, CANCEL, INVITE, and OPTIONS from "o" for optional to "c" for conditional, depending on whether a route set has been defined for the dialog or the response code. o Updated Figure 5; adding missing label on "calling" to "completed" transition. o Fixed errored transport example from Section 19.2.1. o Clarified that 17.2.3 and 17.1.3 are rules that define retransmissions. o fixed reported bugs in bnf (missing productions, bad tex markup), etc. Added new SWS production to have an LWS which allows zero spaces, and used that With any separators. Removed the # rule. o ACK for non-2xx has to have the same Route as the request its acknowledging. The text formerly said that the ACK MUST NOT contain Route, this has now radically changed to MUST have Route if the request its cancelling had one. o Clarified that stateless proxies apply Route processing logic to CANCEL requests. o Emphasized that escaping in the hostname portion of SIP URIs is not currently allowed. Various Authors [Page 261] Internet Draft SIP January 28, 2002 o Added discussion on when configuration changes affect the ability of a proxy to forward requests stateful or statelessly. o Explicitly stated that a proxy may add a Record-Route header field value to any request. o Added discussion on the use of To tags in hop-hop responses at a proxy. o Relaxed text concerning proxies forwarding CANCELs when a matching response context can't be found to allow the CANCEL to be processed statefully. o Changed references to "short" form of SIP headers to "compact" form. o Changed Date example to a valid date. o Clarified how ACK gets from transport to UAS core. o Adding missing "SIP/2.0" to first REGISTER in the examples section. o Fixed bug in 17.2.3 which said that an ACK matched a server transaction if the CSeq method (not number) matched that of the INVITE. It should be the reverse; number, not method. o Fixed bug in 22.15 where it said Content-Length instead of Content-Type. o Incorporated draft-ietf-sip-100rel-04 into bis. o Reliability of provisional responses now only defined for provisional responses to INVITE, although extension methods can allow its usage. This is because PRACK needs to be sent within the context of a dialog, and only responses to INVITE establish dialogs. o Can no longer send a reliable provisional response after a final response; its not compatible with the transaction machines, which generally assume no provisionals after a final. o Proxy behavior for reliable provisional responses no longer defined separately; the spec states that it simply acts as a uas. Various Authors [Page 262] Internet Draft SIP January 28, 2002 o Scope of Record-Route header fields for a reliable provisional response is now the dialog rather than the particular request. o Example PRACK flows were lost when incorporating into bis. o Formal IANA registration of "100rel" option tag. o If reliable provisional response gets no PRACK after 32*T1, UAS sends 5xx to original request. o Recommended UA behavior for caching credentials. o Included guidelines for devices presenting pre-configured credentials vs. prompting end users to provide credentials for a specific realm. o Added section on stateless UAS Behavior, clarifying secure handling of unauthenticated requests to prevent potential DoS threat. o Provided motivation for aggregation of challenges in the Security Considerations, and made the behavioral language there more specific. o Provided guidelines for the construction of realm strings for authentication. o Changed concept of protection domain for SIP so that it is no longer defined by both a Request-URI and a realm; it is now only defined by a realm. o Put in some text encouraging UACs not to resubmit rejected credentials when re-challenged. o Added falsification of source IP address to the Via denial of service attack case. o Provided canonical MD5 hash for an empty message body to be used in Digest integrity calculation. o Added security considerations for the CANCEL and ACK methods. o Deprecated and removed Basic auth scheme. Proxies MUST NOT accept or request Basic. o Strengthened language regarding the sending of the "qop" parameter; receipt of cnonce is based on "qop". Various Authors [Page 263] Internet Draft SIP January 28, 2002 o Clarified the construction the URI in the Request-URI of REGISTER requests. o Noted that registrars SHOULD provide Date headers in 200 (OK) responses to REGISTER, and that clients can use these Dates to set their internal clocks. o Processing of REGISTERs at a registrar now must be with atomicity and isolation. o Registrars now MUST process Require headers. o Clarified CSeq increment over REGISTER messages for the same Call-ID, and necessity of tracking Call-IDs and CSeqs for contact addresses by a registrar o Added registrar-side handling for Contact: * Expires: 0 o Added description generalizing processing of OPTIONS responses to include proxies as well as UAS. Included language describing use of Max-Forwards as a SIP capabilities traceroute. Described construction of a Request-URI for an OPTIONS sent to a proxy. o Defined "Not Applicable" in Tables 2 and 3 to mean that the header field is undefined and should be ignored if present. o Removed old references to general headers in Table 3. o Allowed a proxy to insert a Max-Forwards header field in Table 2. Also added description of the use of the header by elements that can not otherwise guarantee loop detection. o Fixed dialog matching reference in 22.37. o Reinforced that all 6xx, including 603 and 606, are only sent if the UAS knows that no other endpoint will accept the call. o Clarified that for 302 responses, the Contact is used just once to recurse a new transaction, unless an Expires header or expires parameter is present. o Clarified that 405 is sent when the server knows the method, Various Authors [Page 264] Internet Draft SIP January 28, 2002 but the method is not allowed for the resource in the Request-URI. 501 is sent when the server has never heard of the method at all. o Included note that no MIME types for message bodies of 3xx responses have been defined. o Stated explicitly in Section 22.10 on Contact the rules for parsing display names, URI and URI parameters, and header parameters. Referenced this text in the sections on To and From header fields. o Corrected references in Timestamp section. o Noted in Via section that the host or network address and port part of the header does not follow the SIP URI syntax; spaces around : are permitted. Also noted that spaces are permitted around /. Modified an example to show this. o Added text to describe the Contact header fields in a 2xx response to an OPTIONS as having redirect semantics. Modified example to show both a SIP and mailto Contact URI. o Added text to describe the use of OPTIONS within a dialog to query a peer for capabilities, and noted that the request has no impact on the dialog. o Added text to 302 (Moved Temporarily) section saying that if a cached Contact URI fails, the request may be retried with the original Request-URI. Removed recursion rules (moved to UA section) and "call" specific language. Specifically stated both proxies and Uas may cache URI for expiration interval. o Added text to 488/606 section to allow SDP message bodies, formatted the same as SDP in 200 (OK) responses to OPTIONS. Removed text on SDP response message bodies from the Warning section. o Outbound server is now called outbound proxy o Clarified that a transaction in the completed state is not "in progress" when it comes to overlapping transactions. o 488 response is used to reject an offer. o Clarified how to reject an offer. o Clarified that requests with To tag outside a dialog may have Various Authors [Page 265] Internet Draft SIP January 28, 2002 been simply missrouted. o General UAS behaviour applies to CANCEL and BYE o Clarified when to use BYE to terminate an early dialog. o Explained when a UAS detects gaps in the Cseq space. o Specified behavior for inclusion of bodies in ACK for non-2xx; MUST be same type as request, or one of the types in Accept if the response was 415. o Updated the default value of timer D to be 32s, instead of T3. o Clarified that RTT estimate of T1 applies to all requests and responses sent to that IP address, and included a discussion of how this is not quite right when there are stateless proxies in the path. o 180 (Ringing) responses for re-INVITEs are not typically useful. o ACKs MUST contain the same credentials as the INVITE. o ACK for non-2xx responses needs to contain the same Route headers as the request. Same reason CANCEL needs to. o Increased minimum timer for holding persistent connections, and clarified the reasoning behind the timer. o Clarified that persistent connections are indexed by address, port, transport, and that ephemeral source ports imply that peering relationships will ususally involve two connections. o Timer T3 no longer used; it was a dangling reference in bis- 05. o Clarified Figure 7 to indicate that 100 is only sent if TU won't respond in 200ms. o Re-added text that said proxies MUST and UA SHOULD support TCP, which somehow got accidentally deleted from bis-05. o Clarified meaning of an empty Accept header field. o Added RFC 2616 security warning about Server header field to both Server and User-Agent header fields. Various Authors [Page 266] Internet Draft SIP January 28, 2002 o Added handling of transport failures to transaction state machines, and added a section for server transactions. o Disallowed port in To/From header URIs. o Allowed password in both To and From header URIs. o Disallowed the method URI parameter in REGISTER and Redirect Contact header URIs. o Absolved proxies from issuing CANCELs based on the Expires header of an INVITE. Included text pointing out that they MAY do so, but it is unnecessary. o Clarified aggregating authentication challenges at a proxy. o Added notice that even though proxies are required to CANCEL outstanding client transactions upon forwarding a final response, an endpoint may still receive multiple 200 (OK) responses to an INVITE. Also noted that future extensions could override the requirement to CANCEL. o Reinforced that proxies must wait for provisional responses before generating CANCEL requests. o Request merging moved to general Ua behaviour section. o Request processing is atomic. o Clarified how to resolve glare conditions. o Added UAs should ignore unknown extension header parameters. o Clarified when quoted string vs. token can be used as a display name. o Explicitly stated that a header parameter name can appear at most once per header field value. o Noted that proxies no longer treat merged requests as an error. o Clarified that proxies can Record-Route header field values to requests already in dialogs to improve robustness, but that chosing not to do so will not normally cause them to be removed from the path. o Clarified that proxies do not remove any received parameters Various Authors [Page 267] Internet Draft SIP January 28, 2002 they may have added to Via header fields when forwarding responses. o Deprecated absolute time in Expires and Retry-After. o Added pointer to what to do with responses that were meant for a proxy o Summarized stateful proxy forwarding behavior with respect to what final responses get forwarded o Clarified that elements on the start line of messages are separated by a single SP character o Explicitly stated that a SIP URI parameter name to occur at most once in a URI. o Changed Table 2 to show Accept, Accept-Encoding, Accept- Language, and Supported as for a 2xx to an OPTIONS as m* o Changed Table 2 to show Content-Length as "t", which is defined to mean that it should be present, but must be present if TCP is used. o Added the notion that registrars that accept registrations on a multicast interface might want to redirect registrations to a unicast interface. o Request merging now a behavior of the UA, rather than the proxy server. o Solidified the circumstances under which UAs should retry rejected requests with the same Call-ID but a different CSeq. o Corrected erroneous statement that contact addresses were not cached across dialogs; now dependent on status code and expiration interval. o Tags are a MUST for non-100 provisionals, a MAY for 100 (Trying). o Discouraged generation of 1xx respones to non-INVITE requests. o Fixed references to Content handling headers in the UA section. o Timestamp headers must be copied from requests into a 100 Trying for RTT calculation. Various Authors [Page 268] Internet Draft SIP January 28, 2002 o Request processing is now said to be atomic. o Potential infinite redirection loop problem fixed; redirect servers MUST NOT send a redirect to the same URI they received in the redirected request. o Further specified which URIs servers can expect to see in Request-URIs of requests (relationship to contact headers). o Defined pre-loaded route headers. o Clarified normative language of Accept-Encoding, Accept- Language, and Content-Disposition in regard to no header being present. o Noted that "transport=TLS" in a SIP URI refers to TLS over TCP. o Refined discussion on forming requests based on a given SIP URI. o Clarified "matching the topmost Via" for stateless proxies. o Added discussion of how proxies respond to transaction failure and notification of state-machine timeouts. o Corrected description of proxy behavior when recursing on 3xx contacts to account for contacts not recursed on (such as contacts containing non-SIP URIs). o Added Reply-To header field. o Clarified that responses to OPTIONS are scoped to the Request-URI of the request. o Added 491 (Request Pending) response code. o Proxies should not remove malformed headers that it doesn't care about when forwarding requests. o Noted that proxies can't generate their own 1xx provisional responses, but they can use a virutual colocated UAS to achieve the same effect. o Two SIP URIs which are identical with the exception of the presence of an maddr parameter in one, and no maddr parameter in the other are not equivalent. Various Authors [Page 269] Internet Draft SIP January 28, 2002 o Modified transaction, UA, and proxy sections so that branch ID is now a unique transaction identifier. Updated all example messages so that UAC insert branch ID, and magic cookie is present in all branch ID values. o CANCELs and ACKs MUST NOT contain Require or Proxy-Require headers. o A UA SHOULD NOT send re-INVITE or BYE upon media failure. o Only SIP URIs can be used as addresses of record in REGISTER requests. o Registrars MUST NOT increase the expiration interval of registrations. Intervals that are too short MAY be rejected with a 423 w/ Min-Expires. o Security Considerations substantially reorganized and expanded. o TLS support for proxy servers, registrars and redirect servers now a MUST. o Minimum ciphersuite for TLS now AES. o S/MIME now slightly more implementable. S/MIME support is now a SHOULD for UAs. o S/MIME now relies on RFC 2633 CMS messages. o Threat models against the SIP protocol are now provided. o Example architectures in which security mechanisms might be used are described. o Limitations of security mechanisms are described. o Added 493 (Undecipherable) response code. o Fixed ACK column in Table 3 entry for Warning. o Added text describing how to recurse on a 3xx as a UAC. o SIP URIs are compared case-sensitive across the userpart, case-insensitive everywhere else. o Proxies strip transport and port when stripping maddr. Various Authors [Page 270] Internet Draft SIP January 28, 2002 o Port and transport apply to maddr when maddr is present in a SIP URI. o Restored record-route example from bis-04. o Reinforced that SIP messages MAY contain binary bodies or body parts. o Added section discussing conversion of tel URLs to SIP URIs, focusing on issues with maintaining equivalence. o Clarified use of transaction key in building values to include in Record-Route values. o Clarified requirements on the inclusion of information in the loop-detection hash used in branch parameters. o Noted in the proxy section that Record-Route values are only valid within the scope of the dialog in which they are provided. o Added definitions for redirect server, recursion, header, message, request, response, and route refresh request. o Placing headers needed by proxies (Via, Route, Record-Route, etc.) at the top of messages is now RECOMMENDED. o Reinforced that proxies processing messages do not fork, even by recursingon returned 3xx responses. o Removed restriction on proxies adding Record-Route to REGISTER requests. Added that registrars ignore Record-Route if it occurs. o Allowed for loose-route policies, capturing use of default outbound proxies as a loose route decision. o The scope of Contact header fields is not limited to the dialog. o Added text saying that when the caller wishes to be anonymous, the URI should be scrambled as well. o Moved 485 response generation from UAS to proxy. o Require MUST only reference standards track RFCs. o Removed requirement on proxies to not forward a request to a Various Authors [Page 271] Internet Draft SIP January 28, 2002 multicast group that had already been visited. o Deprecated loop-detection. Made Max-Forwards mandatory with an initial value of 70. Proxies insert a Max-Forwards of 70 if they find the header missing. o Placed HTTP Digest and S/MIME in sections independent of the security Considerations. o Added 416 (Unsupported URI Scheme) and discussion on its handling. Added guidance on how a UAC would select the URI in the To/Request-URI based on user input. o Noted that BYE without tags is now rejected, which is a backwards compatibility break with RFC 2543. o Reference offer-answer for formatting of SDP in OPTIONS response, 488, 606. o Timer C now managed by the TU. Proxies have a minimum of 3 minutes, but it is extended through provisional responses. o Proxies can go stateless mid-transaction if they didn't do anything that would have otherwise prevented them from being stateless in the first place. 36 Acknowledgments We wish to thank the members of the IETF MMUSIC and SIP WGs for their comments and suggestions. Detailed comments were provided by Brian Bidulock, Jim Buller, Neil Deason, Dave Devanathan, C‰dric Fluckiger, Yaron Goland, Bernie H÷neisen, Phil Hoffer, Christian Huitema, Jean Jervis, Gadi Karmi, Peter Kjellerstedt, Anders Kristensen, Jonathan Lennox, Gethin Liddell, Alison Mankin, Keith Moore, Vern Paxson, Moshe J. Sambol, Chip Sharp, Igor Slepchin, Eric Tremblay., and Rick Workman. Brian Rosen provided the compiled BNF. This work is based, inter alia, on [43,44]. 37 Authors' Addresses Authors addresses are listed alphabetically for the editors, the writers, and then the original authors of RFC 2543. All listed authors actively contributed large amounts of text to this document. Jonathan Rosenberg Various Authors [Page 272] Internet Draft SIP January 28, 2002 dynamicsoft 72 Eagle Rock Ave East Hanover, NJ 07936 USA electronic mail: jdrosen@dynamicsoft.com Henning Schulzrinne Dept. of Computer Science Columbia University 1214 Amsterdam Avenue New York, NY 10027 USA electronic mail: schulzrinne@cs.columbia.edu Gonzalo Camarillo Ericsson Advanced Signalling Research Lab. FIN-02420 Jorvas Finland electronic mail: Gonzalo.Camarillo@ericsson.com Alan Johnston WorldCom 100 South 4th Street St. Louis, MO 63102 USA electronic mail: alan.johnston@wcom.com Jon Peterson NeuStar, Inc 1800 Sutter Street, Suite 570 Concord, CA 94520 USA electronic mail: jon.peterson@neustar.com Robert Sparks dynamicsoft, Inc. 5100 Tennyson Parkway Suite 1200 Plano, Texas 75024 USA electronic mail: rsparks@dynamicsoft.com Mark Handley ACIRI electronic mail: mjh@aciri.org Eve Schooler Various Authors [Page 273] Internet Draft SIP January 28, 2002 Computer Science Department 256-80 California Institute of Technology Pasadena, CA 91125 USA electronic mail: schooler@cs.caltech.edu 38 Bibliography [1] R. Pandya, "Emerging mobile and personal communication systems," IEEE Communications Magazine , Vol. 33, pp. 44--52, June 1995. [2] H. Schulzrinne, S. Casner, R. Frederick, and V. 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Schulzrinne, "RTP profile for audio and video conferences with minimal control," Request for Comments 1890, Internet Engineering Task Force, Jan. 1996. [42] R. Hinden, B. Carpenter, and L. Masinter, "Format for literal IPv6 addresses in URL's," Request for Comments 2732, Internet Engineering Task Force, Dec. 1999. [43] E. M. Schooler, "Case study: multimedia conference control in a packet-switched teleconferencing system," Journal of Internetworking: Research and Experience , Vol. 4, pp. 99--120, June 1993. ISI reprint series ISI/RS-93-359. [44] H. Schulzrinne, "Personal mobility for multimedia services in the Internet," in European Workshop on Interactive Distributed Multimedia Systems and Services (IDMS) , (Berlin, Germany), Mar. 1996. Full Copyright Statement Copyright (c) The Internet Society (2002). All Rights Reserved. This document and translations of it may be copied and furnished to others, and derivative works that comment on or otherwise explain it or assist in its implementation may be prepared, copied, published and distributed, in whole or in part, without restriction of any kind, provided that the above copyright notice and this paragraph are included on all such copies and derivative works. However, this document itself may not be modified in any way, such as by removing Various Authors [Page 277] Internet Draft SIP January 28, 2002 the copyright notice or references to the Internet Society or other Internet organizations, except as needed for the purpose of developing Internet standards in which case the procedures for copyrights defined in the Internet Standards process must be followed, or as required to translate it into languages other than English. The limited permissions granted above are perpetual and will not be revoked by the Internet Society or its successors or assigns. This document and the information contained herein is provided on an "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. Various Authors [Page 278]