Impedance Mismatch

marc@petit-huguenin.org

Cisco Systems

7200-12 Kit Creek Road Research Triangle Park NC 27709 US gsalguei@cisco.com

RAI AVTCORE This document defines how Datagram Transport Layer Security (DTLS), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Session Traversal Utilities for NAT (STUN), and Traversal Using Relays around NAT (TURN) packets are multiplexed on a single receiving socket. It overrides the guidance from SRTP Extension for DTLS, which suffered from three issues described and fixed in this document.

Section 5.1.2 of Secure Real-time Transport Protocol (SRTP) Extension for DTLS defines a scheme for a Real-time Transport Protocol (RTP) receiver to demultiplex Datagram Transport Layer Security (DTLS), Session Traversal Utilities for NAT (STUN) and Secure Real-time Transport Protocol (SRTP)/Secure Real-time Transport Control Protocol (SRTCP) packets that are arriving on the RTP port. Unfortunately, this demultiplexing scheme has created problematic issues: It implicitly allocated codepoints for new STUN methods without an IANA registry reflecting these new allocations.It implicitly allocated codepoints for new Transport Layer Security (TLS) ContentTypes without an IANA registry reflecting these new allocations.It did not take into account the fact that the Traversal Using Relays around NAT (TURN) usage of STUN can create TURN channels that also need to be demultiplexed with the other packet types explicitly mentioned in Section 5.1.2 of RFC 5764. Having overlapping ranges between different IANA registries becomes an issue when a new codepoint is allocated in one of these registries without carefully anyalyzing the impact it could have on the other registries when that codepoint is demultiplexed. Even if a codepoint is not initially thought to be useful in an RFC 5764 implementation, the respective IANA registry expert should at least raise a flag when the allocated codepoint irrevocably prevents multiplexing. The first goal of this document is to make sure that future allocations in any of the affected protocols are done with the full knowledge of their impact on multiplexing. This is achieved by modifying the IANA registries with instructions for coordination between the protocols at risk. A second goal is to permit the addition of new protocols to the list of existing multiplexed protocols in a manner that does not break existing implementations. The flaws in the demultiplexing scheme were unavoidably inherited by other documents, such as and . So in addition, these and any other affected documents will need to be corrected with the updates this document provides.

The demultiplexing scheme in states that the receiver can identify the packet type by looking at the first byte. If the value of this first byte is 0 or 1, the packet is identified to be STUN. The problem that arises as a result of this implicit allocation is that this restricts the codepoints for STUN methods (as described in Section 18.1 of ) to values between 0x000 and 0x07F, which in turn reduces the number of possible STUN method codepoints assigned by IETF Review (i.e., the range from (0x000 - 0x7FF) from 2048 to only 128 and eliminating the possibility of having STUN method codepoints assigned by Designated Expert (i.e., the range 0x800 - 0xFFF). To preserve the Designated Expert range, this document allocates the value 2 and 3 to also identify STUN methods. The IANA Registry for STUN methods is modified to mark the codepoints from 0x100 to 0xFFF as Reserved. These codepoints can still be allocated, but require IETF Review with a document that will properly evaluate the risk of an assignment overlapping with other registries. In addition, this document also updates the IANA registry such that the STUN method codepoints assigned in the 0x080-0x0FF range are also assigned via Designated Expert. The proposed changes to the STUN Method Registry are: OLD:

0x000-0x7FF IETF Review 0x800-0xFFF Designated Expert

NEW:

0x000-0x07F IETF Review 0x080-0x0FF Designated Expert 0x100-0xFFF Reserved

The demultiplexing scheme in dictates that if the value of the first byte is between 20 and 63 (inclusive), then the packet is identified to be DTLS. The problem that arises is that this restricts the TLS ContentType codepoints (as defined in Section 12 of ) to this range, and by extension implicitly allocates ContentType codepoints 0 to 19 and 64 to 255. With respect to TLS packet identification, this document simply explicitly reserves the codepoints from 0 to 19 and from 64 to 255. These codepoints can still be allocated, but require Standards Action with a document that will properly evaluate the risk of an assignment overlapping with other registries. The proposed changes to the TLS ContentTypes Registry are: OLD:

0-19 Unassigned 20 change_cipher_spec 21 alert 22 handshake 23 application_data 24 heartbeat 25-255 Unassigned

NEW:

0-19 Reserved (Requires coordination, see RFCXXXX) 20 change_cipher_spec 21 alert 22 handshake 23 application_data 24 heartbeat 25-63 Unassigned 64-255 Reserved (Requires coordination, see RFCXXXX)

When used with ICE, an RFC 5764 implementation can receive packets on the same socket from three different paths, as shown in : Directly from the sourceThrough a NATRelayed by a TURN server

+------+ | TURN |<------------------------+ +------+ | | | | +-------------------------+ | | | | | v v | | NAT ----------- | | | | +---------------------+ | | | | | | | | v v v | | | +----------+ +----------+ | RFC 5764 | | RFC 5764 | +----------+ +----------+

Even if the ICE algorithm succeeded in selecting a non-relayed path, it is still possible to receive data from the TURN server. For instance, when ICE is used with aggressive nomination the media path can quickly change until it stabilizes. Also, freeing ICE candidates is optional, so the TURN server can restart forwarding STUN connectivity checks during an ICE restart. TURN channels are an optimization where data packets are exchanged with a 4-byte prefix, instead of the standard 36-byte STUN overhead (see Section 2.5 of ). The problem is that the RFC 5764 demultiplexing scheme does not define what to do with packets received over a TURN channel since these packets will start with a first byte whose value will be between 64 and 127 (inclusive). If the TURN server was instructed to send data over a TURN channel, then the current RFC 5764 demultiplexing scheme will reject these packets. Current implementations violate RFC 5764 for values 64 to 127 (inclusive) and they instead parse packets with such values as TURN. In order to prevent future documents from assigning values from the unused range to a new protocol, this document modifies the RFC 5764 demultiplexing algorithm to properly account for TURN channels by allocating the values from 64 to 79 for this purpose.

The key words "MUST", "MUST NOT", "REQUIRED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in when they appear in ALL CAPS. When these words are not in ALL CAPS (such as "must" or "Must"), they have their usual English meanings, and are not to be interpreted as RFC 2119 key words.

This document updates the text in Section 5.1.2 of as follows: OLD TEXT The process for demultiplexing a packet is as follows. The receiver looks at the first byte of the packet. If the value of this byte is 0 or 1, then the packet is STUN. If the value is in between 128 and 191 (inclusive), then the packet is RTP (or RTCP, if both RTCP and RTP are being multiplexed over the same destination port). If the value is between 20 and 63 (inclusive), the packet is DTLS. This process is summarized in Figure 3.

+----------------+ | 127 < B < 192 -+--> forward to RTP | | packet --> | 19 < B < 64 -+--> forward to DTLS | | | B < 2 -+--> forward to STUN +----------------+ Figure 3: The DTLS-SRTP receiver's packet demultiplexing algorithm. Here the field B denotes the leading byte of the packet.

END OLD TEXT NEW TEXT The process for demultiplexing a packet is as follows. The receiver looks at the first byte of the packet. If the value of this byte is in between 0 and 3 (inclusive), then the packet is STUN. Then if the value is between 20 and 63 (inclusive), the packet is DTLS. Then if the value is between 64 and 79 (inclusive), the packet is TURN Channel. Then if the value is in between 128 and 191 (inclusive), then the packet is RTP (or RTCP, if both RTCP and RTP are being multiplexed over the same destination port). Else if the value does not match any known range then the packet MUST be dropped and an alert MAY be logged. This process is summarized in Figure 3.

+----------------+ | [0..3] -+--> forward to STUN | | packet --> | [20..63] -+--> forward to DTLS | | | [64..79] -+--> forward to TURN Channel | | | [128..191] -+--> forward to RTP +----------------+ Figure 3: The DTLS-SRTP receiver's packet demultiplexing algorithm.

END NEW TEXT

[[Note to RFC Editor: Please remove this section and the reference to before publication.]] This section records the status of known implementations of the protocol defined by this specification at the time of posting of this Internet-Draft, and is based on a proposal described in . The description of implementations in this section is intended to assist the IETF in its decision processes in progressing drafts to RFCs. Please note that the listing of any individual implementation here does not imply endorsement by the IETF. Furthermore, no effort has been spent to verify the information presented here that was supplied by IETF contributors. This is not intended as, and must not be construed to be, a catalog of available implementations or their features. Readers are advised to note that other implementations may exist. According to , "this will allow reviewers and working groups to assign due consideration to documents that have the benefit of running code, which may serve as evidence of valuable experimentation and feedback that have made the implemented protocols more mature. It is up to the individual working groups to use this information as they see fit". Note that there is currently no implementation declared in this section, but the intent is to add RFC 6982 templates here from implementers that support the modifications in this document.

This document updates existing IANA registries, adds a new range for TURN channels in the demuxing algorithm, and madates an ascending order for testing the ranges in the demuxing algorithm. These modifications do not introduce any specific security considerations beyond those detailed in .

This specification contains the registration information for reserved STUN Methods codepoints, as explained in and in accordance with the procedures defined in Section 18.1 of . 0x100-0xFFFReserved (MUST be allocated with IETF Review. For DTLS-SRTP multiplexing collision avoidance see RFC XXXX)RFC5764, RFCXXXX This specification also reassigns the ranges in the STUN Methods Registry as follow: 0x000-0x07FIETF Review 0x080-0x0FFDesignated Expert

This specification contains the registration information for reserved TLS ContentType codepoints, as explained in and in accordance with the procedures defined in Section 12 of . 0-19Reserved (MUST be allocated with Standards Action. For DTLS-SRTP multiplexing collision avoidance see RFC XXXX)N/ARFC5764, RFCXXXX 64-255Reserved (MUST be allocated with Standards Action. For DTLS-SRTP multiplexing collision avoidance see RFC XXXX)N/ARFC5764, RFCXXXX

This specification contains the registration information for reserved TURN Channel Numbers codepoints, as explained in and in accordance with the procedures defined in Section 18 of . 0x5000-0xFFFFReserved (For DTLS-SRTP multiplexing collision avoidance see RFC XXXX)RFCXXXX [RFC EDITOR NOTE: Please replace RFCXXXX with the RFC number of this document.]

The implicit STUN Method codepoint allocations problem was first reported by Martin Thomson in the RTCWEB mailing-list and discussed further with Magnus Westerlund. Thanks to Simon Perreault, Colton Shields, Cullen Jennings, Colin Perkins, Magnus Westerlund, Paul Jones, Jonathan Lennox, Varun Singh and Justin Uberti for the comments, suggestions, and questions that helped improve this document.

In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of- service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes. There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously. [STANDARDS-TRACK] This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP). [STANDARDS-TRACK] This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based multimedia sessions established with the offer/answer model. This protocol is called Interactive Connectivity Establishment (ICE). ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN). ICE can be used by any protocol utilizing the offer/answer model, such as the Session Initiation Protocol (SIP). [STANDARDS-TRACK] This document specifies Version 1.2 of the Transport Layer Security (TLS) protocol. The TLS protocol provides communications security over the Internet. The protocol allows client/server applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery. [STANDARDS-TRACK] Session Traversal Utilities for NAT (STUN) is a protocol that serves as a tool for other protocols in dealing with Network Address Translator (NAT) traversal. It can be used by an endpoint to determine the IP address and port allocated to it by a NAT. It can also be used to check connectivity between two endpoints, and as a keep-alive protocol to maintain NAT bindings. STUN works with many existing NATs, and does not require any special behavior from them. STUN is not a NAT traversal solution by itself. Rather, it is a tool to be used in the context of a NAT traversal solution. This is an important change from the previous version of this specification (RFC 3489), which presented STUN as a complete solution. This document obsoletes RFC 3489. [STANDARDS-TRACK] This document describes a Datagram Transport Layer Security (DTLS) extension to establish keys for Secure RTP (SRTP) and Secure RTP Control Protocol (SRTCP) flows. DTLS keying happens on the media path, independent of any out-of-band signalling channel present. [STANDARDS-TRACK] If a host is located behind a NAT, then in certain situations it can be impossible for that host to communicate directly with other hosts (peers). In these situations, it is necessary for the host to use the services of an intermediate node that acts as a communication relay. This specification defines a protocol, called TURN (Traversal Using Relays around NAT), that allows the host to control the operation of the relay and to exchange packets with its peers using the relay. TURN differs from some other relay control protocols in that it allows a client to communicate with multiple peers using a single relay address. [STANDARDS-TRACK] This document specifies version 1.2 of the Datagram Transport Layer Security (DTLS) protocol. The DTLS protocol provides communications privacy for datagram protocols. The protocol allows client/server applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery. The DTLS protocol is based on the Transport Layer Security (TLS) protocol and provides equivalent security guarantees. Datagram semantics of the underlying transport are preserved by the DTLS protocol. This document updates DTLS 1.0 to work with TLS version 1.2. [STANDARDS-TRACK] This document describes a simple process that allows authors of Internet-Drafts to record the status of known implementations by including an Implementation Status section. This will allow reviewers and working groups to assign due consideration to documents that have the benefit of running code, which may serve as evidence of valuable experimentation and feedback that have made the implemented protocols more mature. The process in this document is offered as an experiment. Authors of Internet-Drafts are encouraged to consider using the process for their documents, and working groups are invited to think about applying the process to all of their protocol specifications. The authors of this document intend to collate experiences with this experiment and to report them to the community. This document specifies how the UDP Transport Layer (UDPTL) protocol, the predominant transport protocol for T.38 fax, can be transported over the Datagram Transport Layer Security (DTLS) protocol, how the usage of UDPTL over DTLS is indicated in the Session Description Protocol (SDP), and how UDPTL over DTLS is negotiated in a session established using the Session Initiation Protocol (SIP). This specification defines a new Session Description Protocol (SDP) Grouping Framework extension, 'BUNDLE'. The extension can be used with the SDP Offer/Answer mechanism to negotiate the usage of a single address:port combination (BUNDLE address) for receiving media, referred to as bundled media, associated with multiple SDP media descriptions ("m=" lines). To assist endpoints in negotiating the use of bundle this specification defines a new SDP attribute, 'bundle-only', which can be used to request that specific media is only used if bundled. There are multiple ways to correlate the bundled RTP packets with the appropriate media descriptions. This specification defines a new Real-time Transport Protocol (RTP) source description (SDES) item and a new RTP header extension that provides an additional way to do this correlation by using them to carry a value that associates the RTP/ RTCP packets with a specific media description.

This section must be removed before publication as an RFC.

Removed Section on "Demultiplexing Algorithm Test Order"Split the Introduction into separate sections

Revert to the RFC 5389, as the stunbis reference was needed only for STUN over SCTP.

Remove any discussion about SCTP until a consensus emerges in TRAM.

Instead of allocating the values that are common on each registry, the specification now only reserves them, giving the possibility to allocate them in case muxing is irrelevant.STUN range is now 0-3m with 2-3 being Designated Expert.TLS ContentType 0-19 and 64-255 are now reserved.Add SCTP over UDP value.If an implementation uses the source IP address/port to separate TURN channels packets then the whole channel numbers are available.If not the prefix is between 64 and 79.First byte test order is now by incremental values, so failure is deterministic.Redraw the demuxing diagram.

Adoption by WG.Add reference to STUNbis.

Change affiliation.