Internet DRAFT - draft-ietf-sip-media-security-requirements
draft-ietf-sip-media-security-requirements
SIP Working Group D. Wing, Ed.
Internet-Draft Cisco
Intended status: Informational S. Fries
Expires: July 13, 2009 Siemens AG
H. Tschofenig
Nokia Siemens Networks
F. Audet
Nortel
January 9, 2009
Requirements and Analysis of Media Security Management Protocols
draft-ietf-sip-media-security-requirements-09
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Abstract
This document describes requirements for a protocol to negotiate a
security context for SIP-signaled SRTP media. In addition to the
natural security requirements, this negotiation protocol must
interoperate well with SIP in certain ways. A number of proposals
have been published and a summary of these proposals is in the
appendix of this document.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Attack Scenarios . . . . . . . . . . . . . . . . . . . . . . . 5
4. Call Scenarios and Requirements Considerations . . . . . . . . 8
4.1. Clipping Media Before Signaling Answer . . . . . . . . . . 8
4.2. Retargeting and Forking . . . . . . . . . . . . . . . . . 9
4.3. Recording . . . . . . . . . . . . . . . . . . . . . . . . 12
4.4. PSTN gateway . . . . . . . . . . . . . . . . . . . . . . . 12
4.5. Call Setup Performance . . . . . . . . . . . . . . . . . . 13
4.6. Transcoding . . . . . . . . . . . . . . . . . . . . . . . 13
4.7. Upgrading to SRTP . . . . . . . . . . . . . . . . . . . . 14
4.8. Interworking with Other Signaling Protocols . . . . . . . 14
4.9. Certificates . . . . . . . . . . . . . . . . . . . . . . . 15
5. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 15
5.1. Key Management Protocol Requirements . . . . . . . . . . . 15
5.2. Security Requirements . . . . . . . . . . . . . . . . . . 17
5.3. Requirements Outside of the Key Management Protocol . . . 19
6. Security Considerations . . . . . . . . . . . . . . . . . . . 20
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 20
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
9.1. Normative References . . . . . . . . . . . . . . . . . . . 20
9.2. Informative References . . . . . . . . . . . . . . . . . . 21
Appendix A. Overview and Evaluation of Existing Keying
Mechanisms . . . . . . . . . . . . . . . . . . . . . 24
A.1. Signaling Path Keying Techniques . . . . . . . . . . . . . 25
A.1.1. MIKEY-NULL . . . . . . . . . . . . . . . . . . . . . . 25
A.1.2. MIKEY-PSK . . . . . . . . . . . . . . . . . . . . . . 25
A.1.3. MIKEY-RSA . . . . . . . . . . . . . . . . . . . . . . 26
A.1.4. MIKEY-RSA-R . . . . . . . . . . . . . . . . . . . . . 26
A.1.5. MIKEY-DHSIGN . . . . . . . . . . . . . . . . . . . . . 26
A.1.6. MIKEY-DHHMAC . . . . . . . . . . . . . . . . . . . . . 26
A.1.7. MIKEY-ECIES and MIKEY-ECMQV (MIKEY-ECC) . . . . . . . 27
A.1.8. Security Descriptions with SIPS . . . . . . . . . . . 27
A.1.9. Security Descriptions with S/MIME . . . . . . . . . . 27
A.1.10. SDP-DH (expired) . . . . . . . . . . . . . . . . . . . 27
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A.1.11. MIKEYv2 in SDP (expired) . . . . . . . . . . . . . . . 27
A.2. Media Path Keying Technique . . . . . . . . . . . . . . . 28
A.2.1. ZRTP . . . . . . . . . . . . . . . . . . . . . . . . . 28
A.3. Signaling and Media Path Keying Techniques . . . . . . . . 28
A.3.1. EKT . . . . . . . . . . . . . . . . . . . . . . . . . 28
A.3.2. DTLS-SRTP . . . . . . . . . . . . . . . . . . . . . . 29
A.3.3. MIKEYv2 Inband (expired) . . . . . . . . . . . . . . . 29
A.4. Evaluation Criteria - SIP . . . . . . . . . . . . . . . . 29
A.4.1. Secure Retargeting and Secure Forking . . . . . . . . 29
A.4.2. Clipping Media Before SDP Answer . . . . . . . . . . . 32
A.4.3. SSRC and ROC . . . . . . . . . . . . . . . . . . . . . 34
A.5. Evaluation Criteria - Security . . . . . . . . . . . . . . 36
A.5.1. Distribution and Validation of Persistent Public
Keys and Certificates . . . . . . . . . . . . . . . . 36
A.5.2. Perfect Forward Secrecy . . . . . . . . . . . . . . . 38
A.5.3. Best Effort Encryption . . . . . . . . . . . . . . . . 40
A.5.4. Upgrading Algorithms . . . . . . . . . . . . . . . . . 41
Appendix B. Out-of-Scope . . . . . . . . . . . . . . . . . . . . 43
B.1. Shared Key Conferencing . . . . . . . . . . . . . . . . . 43
Appendix C. Requirement renumbering in -02 . . . . . . . . . . . 44
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 46
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1. Introduction
The work on media security started when the Session Initiation
Protocol (SIP) was still in its infancy. With the increased SIP
deployment and the availability of new SIP extensions and related
protocols, the need for end-to-end security was re-evaluated. The
procedure of re-evaluating prior protocol work and design decisions
is not an uncommon strategy and, to some extent, considered necessary
to ensure that the developed protocols indeed meet the previously
envisioned needs for the users on the Internet.
This document summarizes media security requirements, i.e.,
requirements for mechanisms that negotiate security context such as
cryptographic keys and parameters for SRTP.
The organization of this document is as follows: Section 2 introduces
terminology, Section 3 describes various attack scenarios against the
signaling path and media path, Section 4 provides an overview about
possible call scenarios, Section 5 lists requirements for media
security. The main part of the document concludes with the security
considerations Section 6, IANA considerations Section 7 and an
acknowledgement section in Section 8. Appendix A lists and compares
available solution proposals. The following Appendix A.4 compares
the different approaches regarding their suitability for the SIP
signaling scenarios described in Appendix A, while Appendix A.5
provides a comparison regarding security aspects. Appendix B lists
non-goals for this document.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119], with the
important qualification that, unless otherwise stated, these terms
apply to the design of the media security key management protocol,
not its implementation or application.
Furthermore, the terminology described in SIP ([RFC3261]) regarding
functions and components are used throughout the document
Additionally, the following items are used in this document:
AOR (Address-of-Record): A SIP or SIPS URI that points to a domain
with a location service that can map the URI to another URI where
the user might be available. Typically, the location service is
populated through registrations. An AOR is frequently thought of
as the "public address" of the user.
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SSRC: The 32-bit value that defines the synchronization source, used
in RTP. These are generally unique, but collisions can occur.
two-time pad: The use of the same key and the same keystream to
encrypt different data. For SRTP, a two-time pad occurs if two
senders are using the same key and the same RTP SSRC value.
Perfect Forward Secrecy (PFS): The property that disclosure of the
long-term secret keying material that is used to derive an agreed
ephemeral key does not compromise the secrecy of agreed keys from
earlier runs.
active adversary: An active adversary is able to alter data
communication to affect its operation (see also [RFC4949]).
passive adversary: A passive adversary is able to learn information
from data communication, but not alter that data communication
(see also[RFC4949]).
signaling path: The signaling path is the route taken by SIP
signaling messages transmitted between the calling and called user
agents. This can be either direct signaling between the calling
and called user agents or, more commonly involves the SIP proxy
servers that were involved in the call setup.
media path: The media path is the route taken by media packets
exchanged by the endpoints. In the simplest case, the endpoints
exchange media directly, and the "media path" is defined by a
quartet of IP addresses and TCP/UDP ports, along with an IP route.
In other cases, this path may include RTP relays, mixers,
transcoders, session border controllers, NATs, or media gateways.
Moreover, as this document discusses requirements for media security,
the nomenclature R-XXX is used to mark requrements, were XXX is the
requirement, which needs to be met.
3. Attack Scenarios
The discussion in this section relates to requirements R-PASS-MEDIA,
R-PASS-SIG, R-ASSOC, R-SIG-MEDIA, R-ACT-ACT, and R-ID-BINDING.
This document classifies adversaries according to their access and
their capabilities. An adversary might have access:
1. only to the media path,
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2. only to the signaling path,
3. to the media path and to the signaling path.
An attacker that can solely be located along the signaling path, and
does not have access to media (item 2), is not considered in this
document.
There are two different types of adversaries, active and passive. An
active adversary may need to be active with regard to the key
exchange relevant information traveling along the media path or
traveling along the signaling path.
Based on their robustness against the adversary capabilities
described above, we can group security mechanisms using the following
labels. This list is generally ordered from easiest to compromise
(at the top) to more difficult to compromise:
+---------------+---------+--------------------------------------+
| SIP signaling | media | abbreviation |
+---------------+---------+--------------------------------------+
| none | passive | no-signaling-passive-media |
| none | active | no-signaling-active-media |
| passive | passive | passive-signaling-passive-media |
| passive | active | passive-signaling-active-media |
| active | passive | active-signaling-passive-media |
| active | active | active-signaling-active-media |
| active | active | active-signaling-active-media-detect |
+---------------+---------+--------------------------------------+
no-signaling-passive-media:
Access to only the media path is sufficient to reveal the content
of the media traffic.
passive-signaling-passive-media:
Passive attack on the signaling and passive attack on the media
path is necessary to reveal the content of the media traffic.
passive-signaling-active-media:
Passive attack on the signaling and active attack on the media
path is necessary to reveal the content of the media traffic.
active-signaling-passive-media:
Active attack on the signaling path and passive attack on the
media path is necessary to reveal the content of the media
traffic.
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no-signaling-active-media:
Active attack on the media path is sufficient to reveal the
content of the media traffic.
active-signaling-active-media:
Active attack on both the signaling path and the media path is
necessary to reveal the content of the media traffic.
active-signaling-active-media-detect:
Active attack on both signaling and media path is necessary to
reveal the content of the media traffic (as with active-signaling-
active-media), and the attack is detectable by protocol messages
exchanged between the end points.
For example, unencrypted RTP is vulnerable to no-signaling-passive-
media.
As another example, Security Descriptions [RFC4568], when protected
by TLS (as it is commonly implemented and deployed), belongs in the
passive-signaling-passive-media category since the adversary needs to
learn the Security Descriptions key by seeing the SIP signaling
message at a SIP proxy (assuming that the adversary is in control of
the SIP proxy). The media traffic can be decrypted using that
learned key.
As another example, DTLS-SRTP falls into active-signaling-active-
media category when DTLS-SRTP is used with a public key based
ciphersuite with self-signed certificates and without SIP-Identity
[RFC4474]. An adversary would have to modify the fingerprint that is
sent along the signaling path and subsequently to modify the
certificates carried in the DTLS handshake that travel along the
media path. If DTLS-SRTP is used with both SIP Identity [RFC4474]
and SIP Connected Identity [RFC4916], the RFC4474 signature protects
both the offer and the answer, and such a system would then belong to
the active-signaling-active-attack-detect category (provided, of
course, the signaling path to the RFC4474 authenticator and verifier
is secured as per RFC4474 and the RFC4474 authenticator and verifier
are behaving as per RFC4474).
The above discussion of DTLS-SRTP demonstrates how a single security
protocol can be in different classes depending on the mode in which
it is operated. Other protocols can achieve similar effect by adding
functions outside of the on-the-wire key management protocol itself.
Although it may be appropriate to deploy lower-classed mechanisms in
some cases, the ultimate security requirement for a media security
negotiation protocol is that it have a mode of operation available in
which is detect-attack, which provides protection against the passive
and active attacks and provides detection of such attacks. That is,
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there must be a way to use the protocol so that an active attack is
required against both the signaling and media paths, and so that such
attacks are detectable by the endpoints.
4. Call Scenarios and Requirements Considerations
The following subsections describe call scenarios that pose the most
challenge to the key management system for media data in cooperation
with SIP signaling.
Throughout the subsections requirements are stated by using the
nomenclature R- to state an explicit requirement. All of the stated
requirements are explanied in detail in section Section 5. The
requirements in section Section 5 are listed according their
association to the key management protocol, to attack scenarios, and
requirements which can be met inside the key management protocol or
outside of the key management protocol.
4.1. Clipping Media Before Signaling Answer
The discussion in this section relates to requirement R-AVOID-
CLIPPING and R-ALLOW-RTP.
Per the SDP Offer/Answer Model [RFC3264],
"Once the offerer has sent the offer, it MUST be prepared to
receive media for any recvonly streams described by that offer.
It MUST be prepared to send and receive media for any sendrecv
streams in the offer, and send media for any sendonly streams in
the offer (of course, it cannot actually send until the peer
provides an answer with the needed address and port information)."
To meet this requirement with SRTP, the offerer needs to know the
SRTP key for arriving media. If either endpoint receives encrypted
media before it has access to the associated SRTP key, it cannot play
the media -- causing clipping.
For key exchange mechanisms that send the answerer's key in SDP, a
SIP provisional response [RFC3261], such as 183 (session progress),
is useful. However, the 183 messages are not reliable unless both
the calling and called end point support PRACK [RFC3262], use TCP
across all SIP proxies, implement Security Preconditions [RFC5027],
or the both ends implement ICE [I-D.ietf-mmusic-ice] and the answerer
implements the reliable provisional response mechanism described in
ICE. Unfortunately, there is not wide deployment of any of these
techniques and there is industry reluctance to require these
techniques to avoid the problems described in this section.
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Note that the receipt of an SDP answer is not always sufficient to
allow media to be played to the offerer. Sometimes, the offerer must
send media in order to open up firewall holes or NAT bindings before
media can be received (for details see
[I-D.ietf-mmusic-media-path-middleboxes]). In this case, even a
solution that makes the key available before the SDP answer arrives
will not help.
Preventing the arrival of early media (i.e., media that arrives at
the SDP offerer before the SDP answer arrives) might obsolete the
R-AVOID-CLIPPING requirement, but at the time of writing such early
media exists in many normal call scenarios.
4.2. Retargeting and Forking
The discussion in this section relates to requirements R-FORK-
RETARGET, R-DISTINCT, R-HERFP, and R-BEST-SECURE.
In SIP, a request sent to a specific AOR but delivered to a different
AOR is called a "retarget". A typical scenario is a "call
forwarding" feature. In Figure 1 Alice sends an INVITE in step 1
that is sent to Bob in step 2. Bob responds with a redirect (SIP
response code 3xx) pointing to Carol in step 3. This redirect
typically does not propagate back to Alice but only goes to a proxy
(i.e., the retargeting proxy) that sends the original INVITE to Carol
in step 4.
+-----+
|Alice|
+--+--+
|
| INVITE (1)
V
+----+----+
| proxy |
++-+-----++
| ^ |
INVITE (2) | | | INVITE (4)
& redirect (3) | | |
V | V
++-++ ++----+
|Bob| |Carol|
+---+ +-----+
Figure 1: Retargeting
Using retargeting might lead to situations where the User Agent
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Client (UAC) does not know where its request will be going. This
might not immediately seem like a serious problem; after all, when
one places a telephone call on the PSTN, one never really knows if it
will be forwarded to a different number, who will pick up the line
when it rings, and so on. However, when considering SIP mechanisms
for authenticating the called party, this function can also make it
difficult to differentiate an intermediary that is behaving
legitimately from an attacker. From this perspective, the main
problems with retargeting are:
Not detectable by the caller: The originating user agent has no
means of anticipating that the condition will arise, nor any means
of determining that it has occurred until the call has already
been set up.
Not preventable by the caller: There is no existing mechanism that
might be employed by the originating user agent in order to
guarantee that the call will not be re-targeted.
The mechanism used by SIP for identifying the calling party is SIP
Identity [RFC4474]. However, due to the nature of retargeting SIP
Identity can only identify the calling party (that is, the party that
initiated the SIP request). Some key exchange mechanisms predate SIP
Identity and include their own identity mechanism (e.g., MIKEY).
However, those built-in identity mechanism also suffer from the SIP
retargeting problem. While Connected Identity [RFC4916] allows
positive identification of the called party, the primary difficulty
still remains that the calling party does not know if a mismatched
called party is legitimate (i.e., due to authorized retargeting) or
illegitimate (i.e., due to unauthorized retargeting by an attacker
above to modify SIP signaling).
In SIP, 'forking' is the delivery of a request to multiple locations.
This happens when a single AOR is registered more than once. An
example of forking is when a user has a desk phone, PC client, and
mobile handset all registered with the same AOR.
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+-----+
|Alice|
+--+--+
|
| INVITE
V
+-----+-----+
| proxy |
++---------++
| |
INVITE | | INVITE
V V
+--+--+ +--+--+
|Bob-1| |Bob-2|
+-----+ +-----+
Figure 2: Forking
With forking, both Bob-1 and Bob-2 might send back SDP answers in SIP
responses. Alice will see those intermediate (18x) and final (200)
responses. It is useful for Alice to be able to associate the SIP
response with the incoming media stream. Although this association
can be done with ICE [I-D.ietf-mmusic-ice], and ICE is useful to make
this association with RTP, it is not desirable to require ICE to
accomplish this association.
Forking and retargeting are often used together. For example, a boss
and secretary might have both phones ring (forking) and rollover to
voice mail if neither phone is answered (retargeting).
To maintain security of the media traffic, only the end point that
answers the call should know the SRTP keys for the session. Forked
and re-targeted calls only reveal sensitive information to non-
responders when the signaling messages contain sensitive information
(e.g., SRTP keys) that is accessible by parties that receive the
offer, but may not respond (i.e., the original recipients in a
retargeted call, or non-answering endpoints in a forked call). For
key exchange mechanisms that do not provide secure forking or secure
retargeting, one workaround is to re-key immediately after forking or
retargeting. However, because the originator may not be aware that
the call forked this mechanism requires rekeying immediately after
every session is established. This doubles the number of messages
processed by the network.
Further compounding this problem is a unique feature of SIP that when
forking is used, there is always only one final error response
delivered to the sender of the request: the forking proxy is
responsible for choosing which final response to choose in the event
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where forking results in multiple final error responses being
received by the forking proxy. This means that if a request is
rejected, say with information that the keying information was
rejected and providing the far end's credentials, it is very possible
that the rejection will never reach the sender. This problem, called
the Heterogeneous Error Response Forking Problem (HERFP) [RFC3326],
is difficult to solve in SIP. Because we expect the HERFP to
continue to be a problem in SIP for the foreseeable future, a media
security system should function even in the presence of HERFP
behavior.
4.3. Recording
The discussion in this section relates to requirement R-RECORDING.
Some business environments, such as stock brokers, banks, and catalog
call centers, require recording calls with customers. This is the
familiar "this call is being recorded for quality purposes" heard
during calls to these sorts of businesses. In these environments,
media recording is typically performed by an intermediate device
(with RTP, this is typically implemented in a 'sniffer').
When performing such call recording with SRTP, the end-to-end
security is compromised. This is unavoidable, but necessary because
the operation of the business requires such recording. It is
desirable that the media security is not unduly compromised by the
media recording. The endpoint within the organization needs to be
informed that there is an intermediate device and needs to cooperate
with that intermediate device.
This scenario does not place a requirement directly on the key
management protocol. The requirement could be met directly by the
key management protocol (e.g., MIKEY-NULL or [RFC4568]) or through an
external out-of-band-mechanism (e.g., [I-D.wing-sipping-srtp-key]).
4.4. PSTN gateway
The discussion in this section relates to requirement R-PSTN.
It is desirable, even when one leg of a call is on the PSTN, that the
IP leg of the call be protected with SRTP.
A typical case of using media security where two entities are having
a VoIP conversation over IP capable networks. However, there are
cases where the other end of the communication is not connected to an
IP capable network. In this kind of setting, there needs to be some
kind of gateway at the edge of the IP network which converts the VoIP
conversation to format understood by the other network. An example
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of such gateway is a PSTN gateway sitting at the edge of IP and PSTN
networks (such as the architecture described in [RFC3372]).
If media security (e.g., SRTP protection) is employed in this kind of
gateway-setting, then media security and the related key management
is terminated at the PSTN gateway. The other network (e.g., PSTN)
may have its own measures to protect the communication, but this
means that from media security point of view the media security is
not employed truely end-to-end between the communicating entities.
4.5. Call Setup Performance
The discussion in this section relates to requirement R-REUSE.
Some devices lack sufficient processing power to perform public key
operations or Diffie-Hellman operations for each call, or prefer to
avoid performing those operations on every call. The ability to re-
use previous public key or Diffie-Hellman operations can vastly
decrease the call setup delay and processing requirements for such
devices.
In certain devices, it can take a second or two to perform a Diffie-
Hellman operation. Examples of these devices include handsets, IP
Multimedia Services Identity Module (ISIMs), and PSTN gateways. PSTN
gateways typically utilize a Digital Signal Processor (DSP) which is
not yet involved with typical DSP operations at the beginning of a
call, thus the DSP could be used to perform the calculation, so as to
avoid having the central host processor perform the calculation.
However, not all PSTN gateways use DSPs (some have only central
processors or their DSPs are incapable of performing the necessary
public key or Diffie-Hellman operation), and handsets lack a
separate, unused processor to perform these operations.
Two scenarios where R-REUSE is useful are calls between an endpoint
and its voicemail server or its PSTN gateway. In those scenarios
calls are made relatively often and it can be useful for the
voicemail server or PSTN gateway to avoid public key operations for
subsequent calls.
Storing keys across sessions often interferes with perfect forward
secrecy (R-PFS).
4.6. Transcoding
The discussion in this section relates to requirement R-TRANSCODER.
In some environments is is necessary for network equipment to
transcode from one codec (e.g., a highly compressed codec which makes
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efficient use of wireless bandwidth) to another codec (e.g., a
standardized codec to a SIP peering interface). With RTP, a
transcoding function can be performed with the combination of a SIP
B2BUA (to modify the SDP) and a processor to perform the transcoding
between the codecs. However, with end-to-end secured SRTP, a
transcoding function implemented the same way is a man in the middle
attack, and the key management system prevents its use.
However, such a network-based transcoder can still be realized with
the cooperation and approval of the endpoint, and can provide end-to-
transcoder and transcoder-to-end security.
4.7. Upgrading to SRTP
The discussion in this section relates to the requirement R-ALLOW-
RTP.
Legitimate RTP media can be sent to an endpoint for announcements,
colorful ringback tones (e.g., music), advertising, or normal call
progress tones. The RTP may be received before an associated SDP
answer. For details on various scenarios, see
[I-D.stucker-sipping-early-media-coping].
While receiving such RTP exposes the calling party to a risk of
receiving malicious RTP from an attacker, SRTP endpoints will need to
receive and play out RTP media in order to be compatible with
deployed systems that send RTP to calling parties.
4.8. Interworking with Other Signaling Protocols
The discussion in this section relates to the requirement R-OTHER-
SIGNALING.
In many environments, some devices are signaled with protocols other
than SIP which do not share SIP's offer/answer model (e.g., [H.248.1]
or do not utilize SDP (e.g., H.323). In other environments, both
endpoints may be SIP, but may use different key management systems
(e.g., one uses MIKEY-RSA, the other MIKEY-RSA-R).
In these environments, it is desirable to have SRTP -- rather than
RTP -- between the two endpoints. It is always possible, although
undesirable, to interwork those disparate signaling systems or
disparate key management systems by decrypting and re-encrypting each
SRTP packet in a device in the middle of the network (often the same
device performing the signaling interworking). This is undesirable
due to the cost and increased attack area, as such an SRTP/SRTP
interworking device is a valuable attack target.
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At the time of this writing, interworking is considered important.
Interworking without decryption/encryption of the SRTP, while useful,
is not yet deemed critical because the scale of such SRTP deployments
is, to date, relatively small.
4.9. Certificates
The discussion in this section relates to R-CERTS.
On the Internet and on some private networks, validating another
peer's certificate is often done through a trust anchor -- a list of
Certificate Authorities that are trusted. It can be difficult or
expensive for a peer to obtain these certificates. In all cases,
both parties to the call would need to trust the same trust anchor
(i.e., "certificate authority"). For these reasons, it is important
that the media plane key management protocol offer a mechanism that
allows end-users who have no prior association to authenticate to
each other without acquiring credentials from a third party trust
point. Note that this does not rule out mechanisms in which servers
have certificates and attest to the identities of end-users.
5. Requirements
This section is divided into several parts: requirements specific to
the key management protocol (Section 5.1), attack scenarios
(Section 5.2), and requirements which can be met inside the key
management protocol or outside of the key management protocol
(Section 5.3).
5.1. Key Management Protocol Requirements
SIP Forking and Retargeting, from Section 4.2:
R-FORK-RETARGET:
The media security key management protocol MUST securely
support forking and retargeting when all endpoints are willing
to use SRTP without causing the call setup to fail. This
requirement means the endpoints that did not answer the call
MUST NOT learn the SRTP keys (in either direction) used by the
answering endpoint.
R-DISTINCT:
The media security key management protocol MUST be capable of
creating distinct, independent cryptographic contexts for each
endpoint in a forked session.
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R-HERFP:
The media security key management protocol MUST function
securely even in the presence of HERFP behavior, i.e., the
rejection of key information does not reach the sender.
Performance considerations:
R-REUSE:
The media security key management protocol MAY support the re-
use of a previously established security context.
Note: re-use of the security context does not imply re-
use of RTP parameters (e.g., payload type or SSRC).
Media considerations:
R-AVOID-CLIPPING:
The media security key management protocol SHOULD avoid
clipping media before SDP answer without requiring Security
Preconditions [RFC5027]. This requirement comes from
Section 4.1.
R-RTP-CHECK:
If SRTP key negotiation is performed over the media path (i.e.,
using the same UDP/TCP ports as media packets), the key
negotiation packets MUST NOT pass the RTP validity check
defined in Appendix A.1 of [RFC3550], so that SRTP negotiation
packets can be differentiated from RTP packets.
R-ASSOC:
The media security key management protocol SHOULD include a
mechanism for associating key management messages with both the
signaling traffic that initiated the session and with protected
media traffic. It is useful to associate key management
messages with call signaling messages, as this allows the SDP
offerer to avoid performing CPU-consuming operations (e.g.,
Diffie-Hellman or public key operations) with attackers that
have not seen the signaling messages.
For example, if using a Diffie-Hellman keying technique with
security preconditions that forks to 20 end points, the call
initiator would get 20 provisional responses containing 20
signed Diffie-Hellman key pairs. Calculating 20 Diffie-Hellman
secrets and validating signatures can be a difficult task for
some devices. Hence, in the case of forking, it is not
desirable to perform a Diffie-Hellman operation with every
party, but rather only with the party that answers the call
(and incur some media clipping). To do this, the signaling and
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media need to be associated so the calling party knows which
key management exchange needs to be completed. This might be
done by using the transport address indicated in the SDP,
although NATs can complicate this association.
Note: due to RTP's design requirements, it is expected
that SRTP receivers will have to perform authentication
of any received SRTP packets.
R-NEGOTIATE:
The media security key management protocol MUST allow a SIP
User Agent to negotiate media security parameters for each
individual session. Such negotiation MUST NOT cause a two-time
pad (Section 9.1 of [RFC3711]).
R-PSTN:
The media security key management protocol MUST support
termination of media security in a PSTN gateway. This
requirement is from Section 4.4.
5.2. Security Requirements
This section describes overall security requirements and specific
requirements from the attack scenarios (Section 3).
Overall security requirements:
R-PFS:
The media security key management protocol MUST be able to
support perfect forward secrecy.
R-COMPUTE:
The media security key management protocol MUST support
offering additional SRTP cipher suites without incurring
significant computational expense.
R-CERTS:
The key management protocol MUST NOT require that end-users
obtain credentials (certificates or private keys) from a third-
party trust anchor.
R-FIPS:
The media security key management protocol SHOULD use
algorithms that allow FIPS 140-2 [FIPS-140-2] certification or
similar country-specific certification (e.g., [AISITSEC]).
The United States Government can only purchase and use crypto
implementations that have been validated by the FIPS-140
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[FIPS-140-2] process:
"The FIPS-140 standard is applicable to all Federal
agencies that use cryptographic-based security systems to
protect sensitive information in computer and
telecommunication systems, including voice systems. The
adoption and use of this standard is available to private
and commercial organizations."
Some commercial organizations, such as banks and defense
contractors, require or prefer equipment which has received the
same validation.
R-DOS:
The media security key management protocol MUST NOT introduce
any new significant denial of service vulnerabilities (e.g.,
the protocol should not request the endpoint to perform CPU-
intensive operations without the client being able to validate
or authorize the request).
R-EXISTING:
The media security key management protocol SHOULD allow
endpoints to authenticate using pre-existing cryptographic
credentials, e.g., certificates or pre-shared keys.
R-AGILITY:
The media security key management protocol MUST provide crypto-
agility, i.e., the ability to adapt to evolving cryptography
and security requirements (update of cryptographic algorithms
without substantial disruption to deployed implementations)
R-DOWNGRADE:
The media security key management protocol MUST protect cipher
suite negotiation against downgrading attacks.
R-PASS-MEDIA:
The media security key management protocol MUST have a mode
which prevents a passive adversary with access to the media
path from gaining access to keying material used to protect
SRTP media packets.
R-PASS-SIG:
The media security key management protocol MUST have a mode in
which it prevents a passive adversary with access to the
signaling path from gaining access to keying material used to
protect SRTP media packets.
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R-SIG-MEDIA:
The media security key management protocol MUST have a mode in
which it defends itself from an attacker that is solely on the
media path and from an attacker that is solely on the signaling
path. A successful attack refers to the ability for the
adversary to obtain keying material to decrypt the SRTP
encrypted media traffic.
R-ID-BINDING:
The media security key management protocol MUST enable the
media security keys to be cryptographically bound to an
identity of the endpoint.
This allows domains to deploy SIP Identity [RFC4474].
R-ACT-ACT:
The media security key management protocol MUST support a mode
of operation that provides active-signaling-active-media-detect
robustness, and MAY support modes of operation that provide
lower levels of robustness (as described in Section 3).
Failing to meet R-ACT-ACT indicates the protocol can not
provide secure end-to-end media.
5.3. Requirements Outside of the Key Management Protocol
The requirements in this section are for an overall VoIP security
system. These requirements can be met within the key management
protocol itself, or can be solved outside of the key management
protocol itself (e.g., solved in SIP or in SDP).
R-BEST-SECURE:
Even when some end points of a forked or retargeted call are
incapable of using SRTP, a solution MUST be described which
allows the establishment of SRTP associations with SRTP-capable
endpoints and / or RTP associations with non-SRTP-capable
endpoints.
R-OTHER-SIGNALING:
A solution SHOULD be able to negotiate keys for SRTP sessions
created via different call signaling protocols (e.g., between
Jabber, SIP, H.323, MGCP).
R-RECORDING:
A solution SHOULD be described which supports recording of
decrypted media. This requirement comes from Section 4.3.
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R-TRANSCODER:
A solution SHOULD be described which supports intermediate
nodes (e.g., transcoders), terminating or processing media,
between the end points.
R-ALLOW-RTP: A solution SHOULD be described which allows RTP media
to be received by the calling party until SRTP has been
negotiated with the answerer, after which SRTP is preferred
over RTP.
6. Security Considerations
This document lists requirements for securing media traffic. As
such, it addresses security throughout the document.
7. IANA Considerations
This document does not require actions by IANA.
8. Acknowledgements
For contributions to the requirements portion of this document, the
authors would like to thank the active participants of the RTPSEC BoF
and on the RTPSEC mailing list, and a special thanks to Steffen Fries
and Dragan Ignjatic for their excellent MIKEY comparison [RFC5197]
document.
The authors would furthermore like to thank the following people for
their review, suggestions, and comments: Flemming Andreasen, Richard
Barnes, Mark Baugher, Wolfgang Buecker, Werner Dittmann, Lakshminath
Dondeti, John Elwell, Martin Euchner, Hans-Heinrich Grusdt, Christer
Holmberg, Guenther Horn, Peter Howard, Leo Huang, Dragan Ignjatic,
Cullen Jennings, Alan Johnston, Vesa Lehtovirta, Matt Lepinski, David
McGrew, David Oran, Colin Perkins, Eric Raymond, Eric Rescorla, Peter
Schneider, Srinath Thiruvengadam, Dave Ward, Dan York, and Phil
Zimmermann.
9. References
9.1. Normative References
[FIPS-140-2]
NIST, "Security Requirements for Cryptographic Modules",
June 2005, <http://csrc.nist.gov/publications/fips/
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fips140-2/fips1402.pdf>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of
Provisional Responses in Session Initiation Protocol
(SIP)", RFC 3262, June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
9.2. Informative References
[AISITSEC]
"Anwendungshinweise und Interpretationen (AIS) zu ITSEC",
January 2002,
<http://www.bsi.de/zertifiz/zert/interpr/aisitsec.htm>.
[H.248.1] ITU, "Gateway control protocol", June 2000,
<http://www.itu.int/rec/T-REC-H.248/e>.
[I-D.baugher-mmusic-sdp-dh]
Baugher, M. and D. McGrew, "Diffie-Hellman Exchanges for
Multimedia Sessions", draft-baugher-mmusic-sdp-dh-00 (work
in progress), February 2006.
[I-D.dondeti-msec-rtpsec-mikeyv2]
Dondeti, L., "MIKEYv2: SRTP Key Management using MIKEY,
revisited", draft-dondeti-msec-rtpsec-mikeyv2-01 (work in
progress), March 2007.
[I-D.fischl-sipping-media-dtls]
Fischl, J., "Datagram Transport Layer Security (DTLS)
Protocol for Protection of Media Traffic Established with
the Session Initiation Protocol",
draft-fischl-sipping-media-dtls-03 (work in progress),
July 2007.
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[I-D.ietf-avt-dtls-srtp]
McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for Secure
Real-time Transport Protocol (SRTP)",
draft-ietf-avt-dtls-srtp-06 (work in progress),
October 2008.
[I-D.ietf-mmusic-ice]
Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols",
draft-ietf-mmusic-ice-19 (work in progress), October 2007.
[I-D.ietf-mmusic-media-path-middleboxes]
Stucker, B. and H. Tschofenig, "Analysis of Middlebox
Interactions for Signaling Protocol Communication along
the Media Path",
draft-ietf-mmusic-media-path-middleboxes-01 (work in
progress), July 2008.
[I-D.ietf-mmusic-sdp-capability-negotiation]
Andreasen, F., "SDP Capability Negotiation",
draft-ietf-mmusic-sdp-capability-negotiation-09 (work in
progress), July 2008.
[I-D.ietf-msec-mikey-ecc]
Milne, A., "ECC Algorithms for MIKEY",
draft-ietf-msec-mikey-ecc-03 (work in progress),
June 2007.
[I-D.ietf-sip-certs]
Jennings, C. and J. Fischl, "Certificate Management
Service for The Session Initiation Protocol (SIP)",
draft-ietf-sip-certs-07 (work in progress), November 2008.
[I-D.ietf-tls-rfc4346-bis]
Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", draft-ietf-tls-rfc4346-bis-10
(work in progress), March 2008.
[I-D.jennings-sipping-multipart]
Wing, D. and C. Jennings, "Session Initiation Protocol
(SIP) Offer/Answer with Multipart Alternative",
draft-jennings-sipping-multipart-02 (work in progress),
March 2006.
[I-D.mcgrew-srtp-ekt]
McGrew, D., "Encrypted Key Transport for Secure RTP",
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draft-mcgrew-srtp-ekt-03 (work in progress), July 2007.
[I-D.stucker-sipping-early-media-coping]
Stucker, B., "Coping with Early Media in the Session
Initiation Protocol (SIP)",
draft-stucker-sipping-early-media-coping-03 (work in
progress), October 2006.
[I-D.wing-sipping-srtp-key]
Wing, D., Audet, F., Fries, S., Tschofenig, H., and A.
Johnston, "Secure Media Recording and Transcoding with the
Session Initiation Protocol",
draft-wing-sipping-srtp-key-04 (work in progress),
October 2008.
[I-D.zimmermann-avt-zrtp]
Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
Path Key Agreement for Secure RTP",
draft-zimmermann-avt-zrtp-11 (work in progress),
November 2008.
[RFC3326] Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason
Header Field for the Session Initiation Protocol (SIP)",
RFC 3326, December 2002.
[RFC3372] Vemuri, A. and J. Peterson, "Session Initiation Protocol
for Telephones (SIP-T): Context and Architectures",
BCP 63, RFC 3372, September 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC4492] Blake-Wilson, S., Bolyard, N., Gupta, V., Hawk, C., and B.
Moeller, "Elliptic Curve Cryptography (ECC) Cipher Suites
for Transport Layer Security (TLS)", RFC 4492, May 2006.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
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[RFC4650] Euchner, M., "HMAC-Authenticated Diffie-Hellman for
Multimedia Internet KEYing (MIKEY)", RFC 4650,
September 2006.
[RFC4738] Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY-
RSA-R: An Additional Mode of Key Distribution in
Multimedia Internet KEYing (MIKEY)", RFC 4738,
November 2006.
[RFC4771] Lehtovirta, V., Naslund, M., and K. Norrman, "Integrity
Transform Carrying Roll-Over Counter for the Secure Real-
time Transport Protocol (SRTP)", RFC 4771, January 2007.
[RFC4916] Elwell, J., "Connected Identity in the Session Initiation
Protocol (SIP)", RFC 4916, June 2007.
[RFC4949] Shirey, R., "Internet Security Glossary, Version 2",
RFC 4949, August 2007.
[RFC5027] Andreasen, F. and D. Wing, "Security Preconditions for
Session Description Protocol (SDP) Media Streams",
RFC 5027, October 2007.
[RFC5197] Fries, S. and D. Ignjatic, "On the Applicability of
Various Multimedia Internet KEYing (MIKEY) Modes and
Extensions", RFC 5197, June 2008.
Appendix A. Overview and Evaluation of Existing Keying Mechanisms
Based on how the SRTP keys are exchanged, each SRTP key exchange
mechanism belongs to one general category:
signaling path:
All the keying is carried in the call signaling (SIP or SDP)
path.
media path:
All the keying is carried in the SRTP/SRTCP media path, and no
signaling whatsoever is carried in the call signaling path.
signaling and media path:
Parts of the keying are carried in the SRTP/SRTCP media path,
and parts are carried in the call signaling (SIP or SDP) path.
One of the significant benefits of SRTP over other end-to-end
encryption mechanisms, such as for example IPsec, is that SRTP is
bandwidth efficient and SRTP retains the header of RTP packets.
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Bandwidth efficiency is vital for VoIP in many scenarios where access
bandwidth is limited or expensive, and retaining the RTP header is
important for troubleshooting packet loss, delay, and jitter.
Related to SRTP's characteristics is a goal that any SRTP keying
mechanism to also be efficient and not cause additional call setup
delay. Contributors to additional call setup delay include network
or database operations: retrieval of certificates and additional SIP
or media path messages, and computational overhead of establishing
keys or validating certificates.
When examining the choice between keying in the signaling path,
keying in the media path, or keying in both paths, it is important to
realize the media path is generally 'faster' than the SIP signaling
path. The SIP signaling path has computational elements involved
which parse and route SIP messages. The media path, on the other
hand, does not normally have computational elements involved, and
even when computational elements such as firewalls are involved, they
cause very little additional delay. Thus, the media path can be
useful for exchanging several messages to establish SRTP keys. A
disadvantage of keying over the media path is that interworking
different key exchange requires the interworking function be in the
media path, rather than just in the signaling path; in practice this
involvement is probably unavoidable anyway.
A.1. Signaling Path Keying Techniques
A.1.1. MIKEY-NULL
MIKEY-NULL [RFC3830] has the offerer indicate the SRTP keys for both
directions. The key is sent unencrypted in SDP, which means the SDP
must be encrypted hop-by-hop (e.g., by using TLS (SIPS)) or end-to-
end (e.g., by using S/MIME).
MIKEY-NULL requires one message from offerer to answerer (half a
round trip), and does not add additional media path messages.
A.1.2. MIKEY-PSK
MIKEY-PSK (pre-shared key) [RFC3830] requires that all endpoints
share one common key. MIKEY-PSK has the offerer encrypt the SRTP
keys for both directions using this pre-shared key.
MIKEY-PSK requires one message from offerer to answerer (half a round
trip), and does not add additional media path messages.
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A.1.3. MIKEY-RSA
MIKEY-RSA [RFC3830] has the offerer encrypt the keys for both
directions using the intended answerer's public key, which is
obtained from a mechanism outside of MIKEY.
MIKEY-RSA requires one message from offerer to answerer (half a round
trip), and does not add additional media path messages. MIKEY-RSA
requires the offerer to obtain the intended answerer's certificate.
A.1.4. MIKEY-RSA-R
MIKEY-RSA-R [RFC4738] is essentially the same as MIKEY-RSA but
reverses the role of the offerer and the answerer with regards to
providing the keys. That is, the answerer encrypts the keys for both
directions using the offerer's public key. Both the offerer and
answerer validate each other's public keys using a standard X.509
validation techniques. MIKEY-RSA-R also enables sending certificates
in the MIKEY message.
MIKEY-RSA-R requires one message from offerer to answer, and one
message from answerer to offerer (full round trip), and does not add
additional media path messages. MIKEY-RSA-R requires the offerer
validate the answerer's certificate.
A.1.5. MIKEY-DHSIGN
In MIKEY-DHSIGN [RFC3830] the offerer and answerer derive the key
from a Diffie-Hellman exchange. In order to prevent an active man-
in-the-middle the DH exchange itself is signed using each endpoint's
private key and the associated public keys are validated using
standard X.509 validation techniques.
MIKEY-DHSIGN requires one message from offerer to answerer, and one
message from answerer to offerer (full round trip), and does not add
additional media path messages. MIKEY-DHSIGN requires the offerer
and answerer to validate each other's certificates. MIKEY-DHSIGN
also enables sending the answerer's certificate in the MIKEY message.
A.1.6. MIKEY-DHHMAC
MIKEY-DHHMAC [RFC4650] uses a pre-shared secret to HMAC the Diffie-
Hellman exchange, essentially combining aspects of MIKEY-PSK with
MIKEY-DHSIGN, but without MIKEY-DHSIGN's need for certificate
authentication.
MIKEY-DHHMAC requires one message from offerer to answerer, and one
message from answerer to offerer (full round trip), and does not add
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additional media path messages.
A.1.7. MIKEY-ECIES and MIKEY-ECMQV (MIKEY-ECC)
ECC Algorithms For MIKEY [I-D.ietf-msec-mikey-ecc] describes how ECC
can be used with MIKEY-RSA (using ECDSA signature) and with MIKEY-
DHSIGN (using a new DH-Group code), and also defines two new ECC-
based algorithms, Elliptic Curve Integrated Encryption Scheme (ECIES)
and Elliptic Curve Menezes-Qu-Vanstone (ECMQV) .
With this proposal, the ECDSA signature, MIKEY-ECIES, and MIKEY-ECMQV
function exactly like MIKEY-RSA, and the new DH-Group code function
exactly like MIKEY-DHSIGN. Therefore these ECC mechanisms are not
discussed separately in this document.
A.1.8. Security Descriptions with SIPS
Security Descriptions [RFC4568] has each side indicate the key it
will use for transmitting SRTP media, and the keys are sent in the
clear in SDP. Security Descriptions relies on hop-by-hop (TLS via
"SIPS:") encryption to protect the keys exchanged in signaling.
Security Descriptions requires one message from offerer to answerer,
and one message from answerer to offerer (full round trip), and does
not add additional media path messages.
A.1.9. Security Descriptions with S/MIME
This keying mechanism is identical to Appendix A.1.8, except that
rather than protecting the signaling with TLS, the entire SDP is
encrypted with S/MIME.
A.1.10. SDP-DH (expired)
SDP Diffie-Hellman [I-D.baugher-mmusic-sdp-dh] exchanges Diffie-
Hellman messages in the signaling path to establish session keys. To
protect against active man-in-the-middle attacks, the Diffie-Hellman
exchange needs to be protected with S/MIME, SIPS, or SIP Identity
[RFC4474] and SIP Conected Identity [RFC4916].
SDP-DH requires one message from offerer to answerer, and one message
from answerer to offerer (full round trip), and does not add
additional media path messages.
A.1.11. MIKEYv2 in SDP (expired)
MIKEYv2 [I-D.dondeti-msec-rtpsec-mikeyv2] adds mode negotiation to
MIKEYv1 and removes the time synchronization requirement. It
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therefore now takes 2 round-trips to complete. In the first round
trip, the communicating parties learn each other's identities, agree
on a MIKEY mode, crypto algorithm, SRTP policy, and exchanges nonces
for replay protection. In the second round trip, they negotiate
unicast and/or group SRTP context for SRTP and/or SRTCP.
Furthemore, MIKEYv2 also defines an in-band negotiation mode as an
alternative to SDP (see Appendix A.3.3).
A.2. Media Path Keying Technique
A.2.1. ZRTP
ZRTP [I-D.zimmermann-avt-zrtp] does not exchange information in the
signaling path (although it's possible for endpoints to exchange a
hash of the ZRTP Hello message with "a=zrtp-hash" in the initial
Offer if sent over an integrity-protected signaling channel. This
provides some useful correlation between the signaling and media
layers). In ZRTP the keys are exchanged entirely in the media path
using a Diffie-Hellman exchange. The advantage to this mechanism is
that the signaling channel is used only for call setup and the media
channel is used to establish an encrypted channel -- much like
encryption devices on the PSTN. ZRTP uses voice authentication of
its Diffie-Hellman exchange by having each person read digits or
words to the other person. Subsequent sessions with the same ZRTP
endpoint can be authenticated using the stored hash of the previously
negotiated key rather than voice authentication. ZRTP uses 4 media
path messages (Hello, Commit, DHPart1, and DHPart2) to establish the
SRTP key, and 3 media path confirmation messages. These initial
messages are all sent as non-RTP packets.
Note that when ZRTP probing is used, unencrypted RTP can be
exchanged until the SRTP keys are established.
A.3. Signaling and Media Path Keying Techniques
A.3.1. EKT
EKT [I-D.mcgrew-srtp-ekt] relies on another SRTP key exchange
protocol, such as Security Descriptions or MIKEY, for bootstrapping.
In the initial phase, each member of a conference uses an SRTP key
exchange protocol to establish a common key encryption key (KEK).
Each member may use the KEK to securely transport its SRTP master key
and current SRTP rollover counter (ROC), via RTCP, to the other
participants in the session.
EKT requires the offerer to send some parameters (EKT_Cipher, KEK,
and security parameter index (SPI)) via the bootstrapping protocol
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such as Security Descriptions or MIKEY. Each answerer sends an SRTCP
message which contains the answerer's SRTP Master Key, rollover
counter, and the SRTP sequence number. Rekeying is done by sending a
new SRTCP message. For reliable transport, multiple RTCP messages
need to be sent.
A.3.2. DTLS-SRTP
DTLS-SRTP [I-D.ietf-avt-dtls-srtp] exchanges public key fingerprints
in SDP [I-D.fischl-sipping-media-dtls] and then establishes a DTLS
session over the media channel. The endpoints use the DTLS handshake
to agree on crypto suites and establish SRTP session keys. SRTP
packets are then exchanged between the endpoints.
DTLS-SRTP requires one message from offerer to answerer (half round
trip), and one message from the answerer to offerer (full round trip)
so the offerer can correlate the SDP answer with the answering
endpoint. DTLS-SRTP uses 4 media path messages to establish the SRTP
key.
This document assumes DTLS will use TLS_RSA_WITH_AES_128_CBC_SHA as
its cipher suite, which is the mandatory-to-implement cipher suite in
TLS [I-D.ietf-tls-rfc4346-bis].
A.3.3. MIKEYv2 Inband (expired)
As defined in Appendix A.1.11, MIKEYv2 also defines an in-band
negotiation mode as an alternative to SDP (see Appendix A.3.3). The
details are not sorted out in the draft yet on what in-band actually
means (i.e., UDP, RTP, RTCP, etc.).
A.4. Evaluation Criteria - SIP
This section considers how each keying mechanism interacts with SIP
features.
A.4.1. Secure Retargeting and Secure Forking
Retargeting and forking of signaling requests is described within
Section 4.2. The following builds upon this description.
The following list compares the behavior of secure forking, answering
association, two-time pads, and secure retargeting for each keying
mechanism.
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MIKEY-NULL Secure Forking: No, all AORs see offerer's and
answerer's keys. Answer is associated with media by the SSRC
in MIKEY. Additionally, a two-time pad occurs if two branches
choose the same 32-bit SSRC and transmit SRTP packets.
Secure Retargeting: No, all targets see offerer's and
answerer's keys. Suffers from retargeting identity problem.
MIKEY-PSK
Secure Forking: No, all AORs see offerer's and answerer's keys.
Answer is associated with media by the SSRC in MIKEY. Note
that all AORs must share the same pre-shared key in order for
forking to work at all with MIKEY-PSK. Additionally, a two-
time pad occurs if two branches choose the same 32-bit SSRC and
transmit SRTP packets.
Secure Retargeting: Not secure. For retargeting to work, the
final target must possess the correct PSK. As this is likely
in scenarios were the call is targeted to another device
belonging to the same user (forking), it is very unlikely that
other users will possess that PSK and be able to successfully
answer that call.
MIKEY-RSA
Secure Forking: No, all AORs see offerer's and answerer's keys.
Answer is associated with media by the SSRC in MIKEY. Note
that all AORs must share the same private key in order for
forking to work at all with MIKEY-RSA. Additionally, a two-
time pad occurs if two branches choose the same 32-bit SSRC and
transmit SRTP packets.
Secure Retargeting: No.
MIKEY-RSA-R
Secure Forking: Yes. Answer is associated with media by the
SSRC in MIKEY.
Secure Retargeting: Yes.
MIKEY-DHSIGN
Secure Forking: Yes, each forked endpoint negotiates unique
keys with the offerer for both directions. Answer is
associated with media by the SSRC in MIKEY.
Secure Retargeting: Yes, each target negotiates unique keys
with the offerer for both directions.
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MIKEYv2 in SDP
The behavior will depend on which mode is picked.
MIKEY-DHHMAC
Secure Forking: Yes, each forked endpoint negotiates unique
keys with the offerer for both directions. Answer is
associated with media by the SSRC in MIKEY.
Secure Retargeting: Yes, each target negotiates unique keys
with the offerer for both directions. Note that for the keys
to be meaningful, it would require the PSK to be the same for
all the potential intermediaries, which would only happen
within a single domain.
Security Descriptions with SIPS
Secure Forking: No. Each forked endpoint sees the offerer's
key. Answer is not associated with media.
Secure Retargeting: No. Each target sees the offerer's key.
Security Descriptions with S/MIME
Secure Forking: No. Each forked endpoint sees the offerer's
key. Answer is not associated with media.
Secure Retargeting: No. Each target sees the offerer's key.
Suffers from retargeting identity problem.
SDP-DH
Secure Forking: Yes. Each forked endpoint calculates a unique
SRTP key. Answer is not associated with media.
Secure Retargeting: Yes. The final target calculates a unique
SRTP key.
ZRTP
Yes. Each forked endpoint calculates a unique SRTP key. With
the "a=zrtp-hash" attribute, the media can be associated with
an answer.
Secure Retargeting: Yes. The final target calculates a unique
SRTP key.
EKT
Secure Forking: Inherited from the bootstrapping mechanism (the
specific MIKEY mode or Security Descriptions). Answer is
associated with media by the SPI in the EKT protocol. Answer
is associated with media by the SPI in the EKT protocol.
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Secure Retargeting: Inherited from the bootstrapping mechanism
(the specific MIKEY mode or Security Descriptions).
DTLS-SRTP
Secure Forking: Yes. Each forked endpoint calculates a unique
SRTP key. Answer is associated with media by the certificate
fingerprint in signaling and certificate in the media path.
Secure Retargeting: Yes. The final target calculates a unique
SRTP key.
MIKEYv2 Inband
The behavior will depend on which mode is picked.
A.4.2. Clipping Media Before SDP Answer
Clipping media before receiving the signaling answer is described
within Section 4.1. The following builds upon this description.
Furthermore, the problem of clipping gets compounded when forking is
used. For example, if using a Diffie-Hellman keying technique with
security preconditions that forks to 20 endpoints, the call initiator
would get 20 provisional responses containing 20 signed Diffie-
Hellman half keys. Calculating 20 DH secrets and validating
signatures can be a difficult task depending on the device
capabilities.
The following list compares the behavior of clipping before SDP
answer for each keying mechanism.
MIKEY-NULL
Not clipped. The offerer provides the answerer's keys.
MIKEY-PSK
Not clipped. The offerer provides the answerer's keys.
MIKEY-RSA
Not clipped. The offerer provides the answerer's keys.
MIKEY-RSA-R
Clipped. The answer contains the answerer's encryption key.
MIKEY-DHSIGN
Clipped. The answer contains the answerer's Diffie-Hellman
response.
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MIKEY-DHHMAC
Clipped. The answer contains the answerer's Diffie-Hellman
response.
MIKEYv2 in SDP
The behavior will depend on which mode is picked.
Security Descriptions with SIPS
Clipped. The answer contains the answerer's encryption key.
Security Descriptions with S/MIME
Clipped. The answer contains the answerer's encryption key.
SDP-DH
Clipped. The answer contains the answerer's Diffie-Hellman
response.
ZRTP
Not clipped because the session intially uses RTP. While RTP
is flowing, both ends negotiate SRTP keys in the media path and
then switch to using SRTP.
EKT
Not clipped, as long as the first RTCP packet (containing the
answerer's key) is not lost in transit. The answerer sends its
encryption key in RTCP, which arrives at the same time (or
before) the first SRTP packet encrypted with that key.
Note: RTCP needs to work, in the answerer-to-offerer
direction, before the offerer can decrypt SRTP media.
DTLS-SRTP
No clipping after the DTLS-SRTP handshake has completed. SRTP
keys are exchanged in the media path. Need to wait for SDP
answer to ensure DTLS-SRTP handshake was done with an
authorized party.
If a middlebox interferes with the media path, there can be
clipping [I-D.ietf-mmusic-media-path-middleboxes].
MIKEYv2 Inband
Not clipped. Keys are exchanged in the media path without
relying on the signaling path.
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A.4.3. SSRC and ROC
In SRTP, a cryptographic context is defined as the SSRC, destination
network address, and destination transport port number. Whereas RTP,
a flow is defined as the destination network address and destination
transport port number. This results in a problem -- how to
communicate the SSRC so that the SSRC can be used for the
cryptographic context.
Two approaches have emerged for this communication. One, used by all
MIKEY modes, is to communicate the SSRCs to the peer in the MIKEY
exchange. Another, used by Security Descriptions, is to apply "late
binding" -- that is, any new packet containing a previously-unseen
SSRC (which arrives at the same destination network address and
destination transport port number) will create a new cryptographic
context. Another approach, common amongst techniques with media-path
SRTP key establishment, is to require a handshake over that media
path before SRTP packets are sent. MIKEY's approach changes RTP's
SSRC collision detection behavior by requiring RTP to pre-establish
the SSRC values for each session.
Another related issue is that SRTP introduces a rollover counter
(ROC), which records how many times the SRTP sequence number has
rolled over. As the sequence number is used for SRTP's default
ciphers, it is important that all endpoints know the value of the
ROC. The ROC starts at 0 at the beginning of a session.
Some keying mechanisms cause a two-time pad to occur if two endpoints
of a forked call have an SSRC collision.
Note: A proposal has been made to send the ROC value on every Nth
SRTP packet[RFC4771]. This proposal has not yet been incorporated
into this document.
The following list examines handling of SSRC and ROC:
MIKEY-NULL
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEY-PSK
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
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MIKEY-RSA
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEY-RSA-R
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEY-DHSIGN
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEY-DHHMAC
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEYv2 in SDP
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
Security Descriptions with SIPS
Neither SSRC nor ROC are signaled. SSRC 'late binding' is
used.
Security Descriptions with S/MIME
Neither SSRC nor ROC are signaled. SSRC 'late binding' is
used.
SDP-DH
Neither SSRC nor ROC are signaled. SSRC 'late binding' is
used.
ZRTP
Neither SSRC nor ROC are signaled. SSRC 'late binding' is
used.
EKT
The SSRC of the SRTCP packet containing an EKT update
corresponds to the SRTP master key and other parameters within
that packet.
DTLS-SRTP
Neither SSRC nor ROC are signaled. SSRC 'late binding' is
used.
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MIKEYv2 Inband
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
A.5. Evaluation Criteria - Security
This section evaluates each keying mechanism on the basis of their
security properties.
A.5.1. Distribution and Validation of Persistent Public Keys and
Certificates
Using persistent public keys for confidentiality and authentication
can introduce requirements for two types of systems, often
implemented using certificates: (1) a system to distribute those
persistent public keys certificates, and (2) a system for validating
those persistent public keys. We refer to the former as a key
distribution system and the latter as an authentication
infrastructure. In many cases, a monolithic public key
infrastructure (PKI) is used for fulfill both of these roles.
However, these functions can be provided by many other systems. For
instance, key distribution may be accomplished by any public
repository of keys. Any system in which the two endpoints have
access to trust anchors and intermediate CA certificates that can be
used to validate other endpoints' certificates (including a system of
self-signed certificates) can be used to support certificate
validation in the below schemes.
With real-time communications it is desirable to avoid fetching or
validating certificates that delay call setup. Rather, it is
preferable to fetch or validate certificates in such a way that call
setup is not delayed. For example, a certificate can be validated
while the phone is ringing or can be validated while ring-back tones
are being played or even while the called party is answering the
phone and saying "hello". Even better is to avoid fetching or
validating persistent public keys at all.
SRTP key exchange mechanisms that require a particular authentication
infrastructure to operate (whether for distribution or validation)
are gated on the deployment of a such an infrastructure available to
both endpoints. This means that no media security is achievable
until such an infrastructure exists. For SIP, something like sip-
certs [I-D.ietf-sip-certs] might be used to obtain the certificate of
a peer.
Note: Even if sip-certs [I-D.ietf-sip-certs] was deployed, the
retargeting problem (Appendix A.4.1) would still prevent
successful deployment of keying techniques which require the
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offerer to obtain the actual target's public key.
The following list compares the requirements introduced by the use of
public-key cryptography in each keying mechanism, both for public key
distribution and for certificate validation.
MIKEY-NULL
Public-key cryptography is not used.
MIKEY-PSK
Public-key cryptography is not used. Rather, all endpoints
must have some way to exchange per-endpoint or per-system pre-
shared keys.
MIKEY-RSA
The offerer obtains the intended answerer's public key before
initiating the call. This public key is used to encrypt the
SRTP keys. There is no defined mechanism for the offerer to
obtain the answerer's public key, although [I-D.ietf-sip-certs]
might be viable in the future.
The offer may also contain a certificate for the offeror, which
would require an authentication infrastructure in order to be
validated by the receiver.
MIKEY-RSA-R
The offer contains the offerer's certificate, and the answer
contains the answerer's certificate. The answerer uses the
public key in the certificate to encrypt the SRTP keys that
will be used by the offerer and the answerer. An
authentication infrastructure is necessary to validate the
certificates.
MIKEY-DHSIGN
An authentication infrastructure is used to authenticate the
public key that is included in the MIKEY message.
MIKEY-DHHMAC
Public-key cryptography is not used. Rather, all endpoints
must have some way to exchange per-endpoint or per-system pre-
shared keys.
MIKEYv2 in SDP
The behavior will depend on which mode is picked.
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Security Descriptions with SIPS
Public-key cryptography is not used.
Security Descriptions with S/MIME
Use of S/MIME requires that the endpoints be able to fetch and
validate certificates for each other. The offerer must obtain
the intended target's certificate and encrypts the SDP offer
with the public key contained in target's certificate. The
answerer must obtain the offerer's certificate and encrypt the
SDP answer with the public key contained in the offerer's
certificate.
SDP-DH
Public-key cryptography is not used.
ZRTP
Public-key cryptography is used (Diffie-Hellman), but without
dependence on persistent public keys. Thus, certificates are
not fetched or validated.
EKT
Public-key cryptography is not used by itself, but might be
used by the EKT bootstrapping keying mechanism (such as certain
MIKEY modes).
DTLS-SRTP
Remote party's certificate is sent in media path, and a
fingerprint of the same certificate is sent in the signaling
path.
MIKEYv2 Inband
The behavior will depend on which mode is picked.
A.5.2. Perfect Forward Secrecy
In the context of SRTP, Perfect Forward Secrecy is the property that
SRTP session keys that protected a previous session are not
compromised if the static keys belonging to the endpoints are
compromised. That is, if someone were to record your encrypted
session content and later acquires either party's private key, that
encrypted session content would be safe from decryption if your key
exchange mechanism had perfect forward secrecy.
The following list describes how each key exchange mechanism provides
PFS.
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MIKEY-NULL
Not applicable; MIKEY-NULL does not have a long-term secret.
MIKEY-PSK
No PFS.
MIKEY-RSA
No PFS.
MIKEY-RSA-R
No PFS.
MIKEY-DHSIGN
PFS is provided with the Diffie-Hellman exchange.
MIKEY-DHHMAC
PFS is provided with the Diffie-Hellman exchange.
MIKEYv2 in SDP
The behavior will depend on which mode is picked.
Security Descriptions with SIPS
Not applicable; Security Descriptions does not have a long-term
secret.
Security Descriptions with S/MIME
Not applicable; Security Descriptions does not have a long-term
secret.
SDP-DH
PFS is provided with the Diffie-Hellman exchange.
ZRTP
PFS is provided with the Diffie-Hellman exchange.
EKT
No PFS.
DTLS-SRTP
PFS is provided if the negotiated cipher suite uses ephemeral
keys (e.g., Diffie-Hellman (DHE_RSA [I-D.ietf-tls-rfc4346-bis])
or Elliptic Curve Diffie-Hellman [RFC4492]).
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MIKEYv2 Inband
The behavior will depend on which mode is picked.
A.5.3. Best Effort Encryption
With best effort encryption, SRTP is used with endpoints that support
SRTP, otherwise RTP is used.
SIP needs a backwards-compatible best effort encryption in order for
SRTP to work successfully with SIP retargeting and forking when there
is a mix of forked or retargeted devices that support SRTP and don't
support SRTP.
Consider the case of Bob, with a phone that only does RTP and a
voice mail system that supports SRTP and RTP. If Alice calls Bob
with an SRTP offer, Bob's RTP-only phone will reject the media
stream (with an empty "m=" line) because Bob's phone doesn't
understand SRTP (RTP/SAVP). Alice's phone will see this rejected
media stream and may terminate the entire call (BYE) and re-
initiate the call as RTP-only, or Alice's phone may decide to
continue with call setup with the SRTP-capable leg (the voice mail
system). If Alice's phone decided to re-initiate the call as RTP-
only, and Bob doesn't answer his phone, Alice will then leave
voice mail using only RTP, rather than SRTP as expected.
Currently, several techniques are commonly considered as candidates
to provide opportunistic encryption:
multipart/alternative
[I-D.jennings-sipping-multipart] describes how to form a
multipart/alternative body part in SIP. The significant issues
with this technique are (1) that multipart MIME is incompatible
with existing SIP proxies, firewalls, Session Border Controllers,
and endpoints and (2) when forking, the Heterogeneous Error
Response Forking Problem (HERFP) [RFC3326] causes problems if such
non-multipart-capable endpoints were involved in the forking.
session attribute
With this technique, the endpoints signal their desire to do SRTP
by signaling RTP (RTP/AVP), and using an attribute ("a=") in the
SDP. This technique is entirely backwards compatible with non-
SRTP-aware endpoints, but doesn't use the RTP/SAVP protocol
registered by SRTP [RFC3711].
SDP Capability Negotiation
SDP Capability Negotiation
[I-D.ietf-mmusic-sdp-capability-negotiation] provides a backwards-
compatible mechanism to allow offering both SRTP and RTP in a
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single offer. This is the preferred technique.
Probing
With this technique, the endpoints first establish an RTP session
using RTP (RTP/AVP). The endpoints send probe messages, over the
media path, to determine if the remote endpoint supports their
keying technique. A disadvantage of probing is an active attacker
can interfere with probes, and until probing completes (and SRTP
is established) the media is in the clear.
The preferred technique, SDP Capability Negotiation
[I-D.ietf-mmusic-sdp-capability-negotiation], can be used with all
key exchange mechanisms. What remains unique is ZRTP, which can also
accomplish its best effort encryption by probing (sending ZRTP
messages over the media path) or by session attribute (see "a=zrtp-
hash" in [I-D.zimmermann-avt-zrtp]). Current implementations of ZRTP
use probing.
A.5.4. Upgrading Algorithms
It is necessary to allow upgrading SRTP encryption and hash
algorithms, as well as upgrading the cryptographic functions used for
the key exchange mechanism. With SIP's offer/answer model, this can
be computionally expensive because the offer needs to contain all
combinations of the key exchange mechanisms (all MIKEY modes,
Security Descriptions) and all SRTP cryptographic suites (AES-128,
AES-256) and all SRTP cryptographic hash functions (SHA-1, SHA-256)
that the offerer supports. In order to do this, the offerer has to
expend CPU resources to build an offer containing all of this
information which becomes computationally prohibitive.
Thus, it is important to keep the offerer's CPU impact fixed so that
offering multiple new SRTP encryption and hash functions incurs no
additional expense.
The following list describes the CPU effort involved in using each
key exchange technique.
MIKEY-NULL
No significant computational expense.
MIKEY-PSK
No significant computational expense.
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MIKEY-RSA
For each offered SRTP crypto suite, the offerer has to perform
RSA operation to encrypt the TGK
MIKEY-RSA-R
For each offered SRTP crypto suite, the offerer has to perform
public key operation to sign the MIKEY message.
MIKEY-DHSIGN
For each offered SRTP crypto suite, the offerer has to perform
Diffie-Hellman operation, and a public key operation to sign
the Diffie-Hellman output.
MIKEY-DHHMAC
For each offered SRTP crypto suite, the offerer has to perform
Diffie-Hellman operation.
MIKEYv2 in SDP
The behavior will depend on which mode is picked.
Security Descriptions with SIPS
No significant computational expense.
Security Descriptions with S/MIME
S/MIME requires the offerer and the answerer to encrypt the SDP
with the other's public key, and to decrypt the received SDP
with their own private key.
SDP-DH
For each offered SRTP crypto suite, the offerer has to perform
a Diffie-Hellman operation.
ZRTP
The offerer has no additional computational expense at all, as
the offer contains no information about ZRTP or might contain
"a=zrtp-hash".
EKT
The offerer's Computational expense depends entirely on the EKT
bootstrapping mechanism selected (one or more MIKEY modes or
Security Descriptions).
DTLS-SRTP
The offerer has no additional computational expense at all, as
the offer contains only a fingerprint of the certificate that
will be presented in the DTLS exchange.
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MIKEYv2 Inband
The behavior will depend on which mode is picked.
Appendix B. Out-of-Scope
The compromise of an endpoint that has access to decrypted media
(e.g., SIP user agent, transcoder, recorder) is out of scope of this
document. Such a compromise might be via privilege escalation,
installation of a virus or trojan horse, or similar attacks.
B.1. Shared Key Conferencing
The consensus on the RTPSEC mailing list was to concentrate on
unicast, point-to-point sessions. Thus, there are no requirements
related to shared key conferencing. This section is retained for
informational purposes.
For efficient scaling, large audio and video conference bridges
operate most efficiently by encrypting the current speaker once and
distributing that stream to the conference attendees. Typically,
inactive participants receive the same streams -- they hear (or see)
the active speaker(s), and the active speakers receive distinct
streams that don't include themselves. In order to maintain
confidentiality of such conferences where listeners share a common
key, all listeners must rekeyed when a listener joins or leaves a
conference.
An important use case for mixers/translators is a conference bridge:
+----+
A --- 1 --->| |
<-- 2 ----| M |
| I |
B --- 3 --->| X |
<-- 4 ----| E |
| R |
C --- 5 --->| |
<-- 6 ----| |
+----+
Figure 3: Centralized Keying
In the figure above, 1, 3, and 5 are RTP media contributions from
Alice, Bob, and Carol, and 2, 4, and 6 are the RTP flows to those
devices carrying the 'mixed' media.
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Several scenarios are possible:
a. Multiple inbound sessions: 1, 3, and 5 are distinct RTP sessions,
b. Multiple outbound sessions: 2, 4, and 6 are distinct RTP
sessions,
c. Single inbound session: 1, 3, and 5 are just different sources
within the same RTP session,
d. Single outbound session: 2, 4, and 6 are different flows of the
same (multi-unicast) RTP session
If there are multiple inbound sessions and multiple outbound sessions
(scenarios a and b), then every keying mechanism behaves as if the
mixer were an end point and can set up a point-to-point secure
session between the participant and the mixer. This is the simplest
situation, but is computationally wasteful, since SRTP processing has
to be done independently for each participant. The use of multiple
inbound sessions (scenario a) doesn't waste computational resources,
though it does consume additional cryptographic context on the mixer
for each participant and has the advantage of data origin
authentication.
To support a single outbound session (scenario d), the mixer has to
dictate its encryption key to the participants. Some keying
mechanisms allow the transmitter to determine its own key, and others
allow the offerer to determine the key for the offerer and answerer.
Depending on how the call is established, the offerer might be a
participant (such as a participant dialing into a conference bridge)
or the offerer might be the mixer (such as a conference bridge
calling a participant). The use of offerless INVITEs may help some
keying mechanisms reverse the role of offerer/answerer. A
difficulty, however, is knowing a priori if the role should be
reversed for a particular call. The significant advantage of a
single outbound session is the number of SRTP encryption operations
remains constant even as the number of participants increases.
However, a disadvantage is that data origin authentication is lost,
allowing any participant to spoof the sender (because all
participants know the sender's SRTP key).
Appendix C. Requirement renumbering in -02
[[RFC Editor: Please delete this section prior to publication.]]
Previous versions of this document used requirement numbers, which
were changed to mnemonics as follows:
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R1 R-FORK-RETARGET
R2 R-BEST-SECURE
R3 R-DISTINCT
R4 R-REUSE; changed from 'MAY' to 'protocol MUST support, and
SHOULD implement'
R5 R-AVOID-CLIPPING
R6 R-PASS-MEDIA
R7 R-PASS-SIG
R8 R-PFS
R9 R-COMPUTE
R10 R-RTP-CHECK
R11 (folded into R4; was reuse previous session)
R12 R-CERTS
R13 R-FIPS
R14 R-ASSOC
R15 R-ALLOW-RTP
R16 R-DOS
R17 R-SIG-MEDIA
R18 R-EXISTING
R19 R-AGILITY
R20 R-DOWNGRADE
R21 R-NEGOTIATE
R23 R-OTHER-SIGNALING
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R23 R-RECORDING (R23 was duplicated in previous versions of the
document)
R24 (deleted; was lawful intercept)
R25 R-TRANSCODER
R26 R-PSTN
R27 R-ID-BINDING
R28 R-ACT-ACT
Authors' Addresses
Dan Wing (editor)
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134
USA
Email: dwing@cisco.com
Steffen Fries
Siemens AG
Otto-Hahn-Ring 6
Munich, Bavaria 81739
Germany
Email: steffen.fries@siemens.com
Hannes Tschofenig
Nokia Siemens Networks
Otto-Hahn-Ring 6
Munich, Bavaria 81739
Germany
Email: Hannes.Tschofenig@nsn.com
URI: http://www.tschofenig.priv.at
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Internet-Draft Media Security Requirements January 2009
Francois Audet
Nortel
4655 Great America Parkway
Santa Clara, CA 95054
USA
Email: audet@nortel.com
Wing, et al. Expires July 13, 2009 [Page 47]
