SIP J. Fischl
Internet-Draft CounterPath Solutions, Inc.
Intended status: Standards Track H. Tschofenig
Expires: May 15, 2008 Nokia Siemens Networks
E. Rescorla
Network Resonance
November 12, 2007
Framework for Establishing an SRTP Security Context using DTLS
draft-ietf-sip-dtls-srtp-framework-00.txt
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Copyright Notice
Copyright (C) The IETF Trust (2007).
Abstract
This document specifies how to use the Session Initiation Protocol
(SIP) to establish an Secure Real-time Transport Protocol (SRTP)
security context using the Datagram Transport Layer Security (DTLS)
protocol. It describes a mechanism of transporting a fingerprint
attribute in the Session Description Protocol (SDP) that identifies
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the key that will be presented during the DTLS handshake. It relies
on the SIP identity mechanism to ensure the integrity of the
fingerprint attribute. The key management exchange travels along the
media path as opposed to the signaling path.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
3. Motivation . . . . . . . . . . . . . . . . . . . . . . . . . . 6
4. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 7
5. Verifying Certificate Integrity . . . . . . . . . . . . . . . 7
6. Miscellaneous Considerations . . . . . . . . . . . . . . . . . 8
6.1. Anonymous Calls . . . . . . . . . . . . . . . . . . . . . 8
6.2. Early Media . . . . . . . . . . . . . . . . . . . . . . . 9
6.3. Forking . . . . . . . . . . . . . . . . . . . . . . . . . 9
6.4. Delayed Offer Calls . . . . . . . . . . . . . . . . . . . 9
6.5. Session Modification . . . . . . . . . . . . . . . . . . . 10
6.6. UDP Payload De-multiplex . . . . . . . . . . . . . . . . . 10
6.7. Rekeying . . . . . . . . . . . . . . . . . . . . . . . . . 10
6.8. Conference Servers and Shared Encryptions Contexts . . . . 11
6.9. Media over SRTP . . . . . . . . . . . . . . . . . . . . . 11
6.10. Best Effort Encryption . . . . . . . . . . . . . . . . . . 11
7. Example Message Flow . . . . . . . . . . . . . . . . . . . . . 11
8. Security Considerations . . . . . . . . . . . . . . . . . . . 16
8.1. UPDATE . . . . . . . . . . . . . . . . . . . . . . . . . . 17
8.2. SIPS . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
8.3. S/MIME . . . . . . . . . . . . . . . . . . . . . . . . . . 18
8.4. Single-sided Verification . . . . . . . . . . . . . . . . 18
8.5. Continuity of Authentication . . . . . . . . . . . . . . . 18
8.6. Short Authentication String . . . . . . . . . . . . . . . 19
8.7. Perfect Forward Secrecy . . . . . . . . . . . . . . . . . 19
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19
10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 20
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
11.1. Normative References . . . . . . . . . . . . . . . . . . . 20
11.2. Informational References . . . . . . . . . . . . . . . . . 21
Appendix A. Requirements Analysis . . . . . . . . . . . . . . . . 23
A.1. Forking and retargeting (R1, R2, R3) . . . . . . . . . . . 23
A.2. Reusage of a Security Context (R4), (R11) . . . . . . . . 23
A.3. Clipping (R5) . . . . . . . . . . . . . . . . . . . . . . 23
A.4. Passive Attacks on the Media Path (R6) . . . . . . . . . . 24
A.5. Passive Attacks on the Signaling Path (R7) . . . . . . . . 24
A.6. Perfect Forward Secrecy (R8) . . . . . . . . . . . . . . . 24
A.7. Algorithm Negotiation (R9) . . . . . . . . . . . . . . . . 24
A.8. RTP Validity Check (R10) . . . . . . . . . . . . . . . . . 24
A.9. 3rd Party Certificates (R12, R18) . . . . . . . . . . . . 24
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A.10. FIPS 140-2 (R13) . . . . . . . . . . . . . . . . . . . . . 24
A.11. Linkage between Keying Exchange and SIP Signaling (R14) . 24
A.12. Start with RTP and Upgrade to SRTP (R15) . . . . . . . . . 25
A.13. Denial of Service Vulnerability (R16) . . . . . . . . . . 25
A.14. Adversary Model (R17) . . . . . . . . . . . . . . . . . . 25
A.15. Crypto-Agility (R19) . . . . . . . . . . . . . . . . . . . 25
A.16. Downgrading Protection (R20) . . . . . . . . . . . . . . . 25
A.17. Media Security Negotation (R21) . . . . . . . . . . . . . 25
A.18. Signaling Protocol Independence (R22) . . . . . . . . . . 25
A.19. Media Recording (R23) . . . . . . . . . . . . . . . . . . 25
A.20. Lawful Interception (R24) . . . . . . . . . . . . . . . . 26
A.21. Interworking with Intermediaries (R25) . . . . . . . . . . 26
A.22. PSTN Gateway Termination (R26) . . . . . . . . . . . . . . 26
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 26
Intellectual Property and Copyright Statements . . . . . . . . . . 28
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1. Introduction
The Session Initiation Protocol (SIP) [RFC3261] and the Session
Description Protocol (SDP) [RFC4566] are used to set up multimedia
sessions or calls. SDP is also used to set up TCP [RFC4145] and
additionally TCP/TLS connections for usage with media sessions
[RFC4572]. The Real-Time Protocol (RTP) [RFC3550] is used to
transmit real time media on top of UDP, TCP [RFC4571], and TLS
[RFC4572]. Datagram TLS [RFC4347] was introduced to allow TLS
functionality to be applied to datagram transport protocols, such as
UDP and DCCP. This draft provides guidelines on how to use and to
support for (a) transmission of media over DTLS and (b) to establish
SRTP security using extensions to DTLS (see
[I-D.ietf-avt-dtls-srtp]).
The goal of this work is to provide a key negotiation technique that
allows encrypted communication between devices with no prior
relationships. It also does not require the devices to trust every
call signaling element that was involved in routing or session setup.
This approach does not require any extra effort by end users and does
not require deployment of certificates to all devices that are signed
by a well-known certificate authority.
The media is transported over a mutually authenticated DTLS session
where both sides have certificates. The certificate fingerprints are
sent in SDP over SIP as part of the offer/answer exchange. The SIP
Identity mechanism [RFC4474] is used to provide integrity for the
fingerprints. It is very important to note that certificates are
being used purely as a carrier for the public keys of the peers.
This is required because DTLS does not have a mode for carrying bare
keys, but it is purely an issue of formatting. The certificates can
be self-signed and completely self-generated. All major TLS stacks
have the capability to generate such certificates on demand.
However, third party certificates MAY also be used for extra
security.
This approach differs from previous attempts to secure media traffic
where the authentication and key exchange protocol (e.g., MIKEY
[RFC3830]) is piggybacked in the signaling message exchange. With
this approach, establishing the protection of the media traffic
between the endpoints is done by the media endpoints without
involving the SIP/SDP communication. It allows RTP and SIP to be
used in the usual manner when there is no encrypted media.
In SIP, typically the caller sends an offer and the callee may
subsequently send one-way media back to the caller before a SIP
answer is received by the caller. The approach in this
specification, where the media key negotiation is decoupled from the
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SIP signaling, allows the early media to be set up before the SIP
answer is received while preserving the important security property
of allowing the media sender to choose some of the keying material
for the media. This also allows the media sessions to be changed,
re-keyed, and otherwise modified after the initial SIP signaling
without any additional SIP signaling.
Design decisions that influence the applicability of this
specification are discussed in Section 3.
2. Overview
Endpoints wishing to set up an RTP media session do so by exchanging
offers and answers in SDP messages over SIP. In a typical use case,
two endpoints would negotiate to transmit audio data over RTP using
the UDP protocol.
Figure 1 shows a typical message exchange in the SIP Trapezoid.
+-----------+ +-----------+
|SIP | SIP/SDP |SIP |
+------>|Proxy |<---------->|Proxy |<------+
| |Server X | (+finger- |Server Y | |
| +-----------+ print, +-----------+ |
| +auth.id.) |
| SIP/SDP SIP/SDP |
| (+fingerprint) (+fingerprint,|
| +auth.id.) |
| |
v v
+-----------+ Datagram TLS +-----------+
|SIP | <---------------------------------> |SIP |
|User Agent | Media |User Agent |
|Alice@X | <=================================> |Bob@Y |
+-----------+ +-----------+
Legend:
<--->: Signaling Traffic
<===>: Data Traffic
Figure 1: DTLS Usage in the SIP Trapezoid
Consider Alice wanting to set up an encrypted audio session with Bob.
Both Bob and Alice could use public-key based authentication in order
to establish a confidentiality protected channel using DTLS.
Since providing mutual authentication between two arbitrary end
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points on the Internet using public key based cryptography tends to
be problematic, we consider more deployment friendly alternatives.
This document uses one approach and several others are discussed in
Section 8.
Alice sends an SDP offer to Bob over SIP. If Alice uses only self-
signed certificates for the communication with Bob, a fingerprint is
included in the SDP offer/answer exchange. This fingerprint is
integrity protected using the identity mechanism defined in
Enhancements for Authenticated Identity Management in SIP [RFC4474].
When Bob receives the offer, Bob establishes a mutually authenticated
DTLS connection with Alice. At this point Bob can begin sending
media to Alice. Once Bob accepts Alice's offer and sends an SDP
answer to Alice, Alice can begin sending confidential media to Bob.
3. Motivation
Although there is already prior work in this area (e.g., Secure
Descriptions for SDP [RFC4568], Key Management Extensions [RFC4567]
combined with MIKEY [RFC3830] for authentication and key exchange),
this specification is motivated as follows:
o TLS will be used to offer security for connection-oriented media.
The design of TLS is well-known and implementations are widely
available.
o This approach deals with forking and early media without requiring
support for PRACK [RFC3262] while preserving the important
security property of allowing the offerer to choose keying
material for encrypting the media.
o The establishment of security protection for the media path is
also provided along the media path and not over the signaling
path. In many deployment scenarios, the signaling and media
traffic travel along a different path through the network.
o This solution works even when the SIP proxies downstream of the
identity service are not trusted. There is no need to reveal keys
in the SIP signaling or in the SDP message exchange. In order for
SDES and MIKEY to provide this security property, they require
distribution of certificates to the endpoints that are signed by
well known certificate authorities. SDES further requires that
the endpoints employ S/MIME to encrypt the keying material.
o In this method, SSRC collisions do not result in any extra SIP
signaling.
o Many SIP endpoints already implement TLS. The changes to existing
SIP and RTP usage are minimal even when DTLS-SRTP
[I-D.ietf-avt-dtls-srtp] is used.
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4. 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].
DTLS/TLS uses the term "session" to refer to a long-lived set of
keying material that spans associations. In this document,
consistent with SIP/SDP usage, we use it to refer to a multimedia
session and use the term "TLS session" to refer to the TLS construct.
We use the term "association" to refer to a particular DTLS
ciphersuite and keying material set. For consistency with other SIP/
SDP usage, we use the term "connection" when what's being referred to
is a multimedia stream that is not specifically DTLS/TLS.
In this document, the term "Mutual DTLS" indicates that both the DTLS
client and server present certificates even if one or both
certificates are self-signed.
5. Verifying Certificate Integrity
The offer/answer model, defined in [RFC3264], is used by protocols
like the Session Initiation Protocol (SIP) [RFC3261] to set up
multimedia sessions. In addition to the usual contents of an SDP
[RFC4566] message, each 'm' line will also contain several attributes
as specified in [I-D.fischl-mmusic-sdp-dtls], [RFC4145] and
[RFC4572].
The endpoint MUST use the setup and connection attributes defined in
[RFC4145]. A setup:active endpoint will act as a DTLS client and a
setup:passive endpoint will act as a DTLS server. The connection
attribute indicates whether or not to reuse an existing DTLS
association.
The endpoint MUST use the certificate fingerprint attribute as
specified in [RFC4572].
The setup:active endpoint establishes a DTLS association with the
setup:passive endpoint [RFC4145]. Typically, the receiver of the SIP
INVITE request containing an offer will take the setup:active role.
The certificate presented during the DTLS handshake MUST match the
fingerprint exchanged via the signaling path in the SDP. The
security properties of this mechanism are described in Section 8.
If the fingerprint does not match the hashed certificate then the
endpoint MUST tear down the media session immediately.
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When an endpoint wishes to set up a secure media session with another
endpoint it sends an offer in a SIP message to the other endpoint.
This offer includes, as part of the SDP payload, the fingerprint of
the certificate that the endpoint wants to use. The SIP message
containing the offer is sent to the offerer's sip proxy over an
integrity protected channel which will add an identity header
according to the procedures outlined in [RFC4474]. When the far
endpoint receives the SIP message it can verify the identity of the
sender using the identity header. Since the identity header is a
digital signature across several SIP headers, in addition to the
bodies of the SIP message, the receiver can also be certain that the
message has not been tampered with after the digital signature was
applied and added to the SIP message.
The far endpoint (answerer) may now establish a mutually
authenticated DTLS association to the offerer. After completing the
DTLS handshake, information about the authenticated identities,
including the certificates, are made available to the endpoint
application. The answerer is then able to verify that the offerer's
certificate used for authentication in the DTLS handshake can be
associated to the certificate fingerprint contained in the offer in
the SDP. At this point the answerer may indicate to the end user
that the media is secured. The offerer may only tentatively accept
the answerer's certificate since it may not yet have the answerer's
certificate fingerprint.
When the answerer accepts the offer, it provides an answer back to
the offerer containing the answerer's certificate fingerprint. At
this point the offerer can definitively accept or reject the peer's
certificate and the offerer can indicate to the end user that the
media is secured.
Note that the entire authentication and key exchange for securing the
media traffic is handled in the media path through DTLS. The
signaling path is only used to verify the peers' certificate
fingerprints.
6. Miscellaneous Considerations
6.1. Anonymous Calls
When making anonymous calls, a new self-signed certificate SHOULD be
used for each call so that the calls can not be correlated as to
being from the same caller. In situations where some degree of
correlation is acceptable, the same certificate SHOULD be used for a
number of calls in order to enable continuity of authentication, see
Section 8.5.
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Additionally, it MUST be ensured that the Privacy header [RFC3325] is
used in conjunction with the SIP identity mechanism to ensure that
the identity of the user is not asserted when enabling anonymous
calls. Furthermore, the content of the subjectAltName attribute
inside the certificate MUST NOT contain information that either
allows correlation or identification of the user that wishes to place
an anonymous call.
6.2. Early Media
If an offer is received by an endpoint that wishes to provide early
media, it MUST take the setup:active role and can immediately
establish a DTLS association with the other endpoint and begin
sending media. The setup:passive endpoint may not yet have validated
the fingerprint of the active endpoint's certificate. The security
aspects of media handling in this situation are discussed in
Section 8.
6.3. Forking
In SIP, it is possible for a request to fork to multiple endpoints.
Each forked request can result in a different answer. Assuming that
the requester provided an offer, each of the answerers' will provide
a unique answer. Each answerer will create a DTLS association with
the offerer. The offerer can then correlate the SDP answer received
in the SIP message by comparing the fingerprint in the answer to the
hashed certificate for each DTLS association.
Note that in the situation where a request forks to multiple
endpoints that all share the same certificate, there is no way for
the caller to correlate the DTLS associations with the SIP dialogs.
Practically, this is not a problem, since the callees will terminate
the unused associations. No new security problem is introduced here
since endpoints which share the same certificate are assumed to
represent the same user.
6.4. Delayed Offer Calls
An endpoint may send a SIP INVITE request with no offer in it. When
this occurs, the receiver(s) of the INVITE will provide the offer in
the response and the originator will provide the answer in the
subsequent ACK request or in the PRACK request [RFC3262] if both
endpoints support reliable provisional responses. In any event, the
active endpoint still establishes the DTLS association with the
passive endpoint as negotiated in the offer/answer exchange.
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6.5. Session Modification
Once an answer is provided to the offerer, either endpoint MAY
request a session modification which MAY include an updated offer.
This session modification can be carried in either an INVITE or
UPDATE request. In this case, it is RECOMMENDED that the offerer
indicate a request to reuse the existing association (using the
connection attribute) as described in Connection-Oriented Media
[RFC4145]. Once the answer is received, the active endpoint will
either reuse the existing association or establish a new one, tearing
down the existing association as soon as the offer/answer exchange is
completed. The exact association/connection reuse behavior is
specified in RFC 4145 [RFC4145].
6.6. UDP Payload De-multiplex
Interactive Connectivity Establishment (ICE), as specified in
[I-D.ietf-mmusic-ice], provides a methodology of allowing
participants in multi-media sessions to verify mutual connectivity.
In order to make ICE work with this specification the endpoints MUST
be able to demultiplex STUN packets from DTLS packets. STUN
[RFC3489] packets MUST NOT be sent over DTLS.
The first byte of a STUN message is 0 or 1 and it is reasonable to
expect it to remain 0 or 1 for the near future. The first byte of a
DTLS packet is "Type" which can currently have values of 20, 21, 22,
and 23 as defined in ContentType declaration in [RFC4346]. It is
reasonable to expect the first byte to remain under 64 and greater
than 1. For RTP the first byte has a value that is 196 or above. A
viable demultiplexing strategy would be to look at the first byte of
the UDP payload and if the value is less than 2, assume STUN, if
greater or equal to 196 assume RTP, otherwise assume DTLS.
6.7. Rekeying
As with TLS, DTLS endpoints can rekey at any time by redoing the DTLS
handshake. While the rekey is under way, the endpoints continue to
use the previously established keying material for usage with DTLS.
Once the new session keys are established the session can switch to
using these and abandon the old keys. This ensures that latency is
not introduced during the rekeying process.
Further considerations regarding rekeying in case the SRTP security
context is established with DTLS can be found in Section 3.7 of
[I-D.ietf-avt-dtls-srtp].
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6.8. Conference Servers and Shared Encryptions Contexts
It has been proposed that conference servers might use the same
encryption context for all of the participants in a conference. The
advantage of this approach is that the conference server only needs
to encrypt the output for all speakers instead of once per
participant.
This shared encryption context approach is not possible under this
specification. However, it is argued that the effort to encrypt each
RTP packet is small compared to the other tasks performed by the
conference server such as the codec processing.
Future extensions such as [I-D.mcgrew-srtp-ekt] could be used to
provide this functionality in concert with the mechanisms described
in this specification.
6.9. Media over SRTP
Because DTLS's data transfer protocol is generic, it is less highly
optimized for use with RTP than is SRTP [RFC3711], which has been
specifically tuned for that purpose. DTLS-SRTP
[I-D.ietf-avt-dtls-srtp], has been defined to provide for the
negotiation of SRTP transport using a DTLS connection, thus allowing
the performance benefits of SRTP with the easy key management of
DTLS. The ability to reuse existing SRTP software and hardware
implementations may in some environments another important motivation
for using DTLS-SRTP instead of RTP over DTLS. Implementations of
this specification SHOULD support DTLS-SRTP [I-D.ietf-avt-dtls-srtp].
6.10. Best Effort Encryption
[I-D.ietf-sip-media-security-requirements] describes a requirement
for best effort encryption where SRTP is used where both endpoints
support it and key negotiation succeeds otherwise RTP is used.
[I-D.ietf-mmusic-sdp-capability-negotiation] describes a mechanism
which can signal both RTP and SRTP as an alternative. RTP is the
default and will be understood by endpoints that do not understand
SRTP or this key exchange mechanism but SRTP is preferred.
7. Example Message Flow
Prior to establishing the session, both Alice and Bob generate self-
signed certificates which are used for a single session or, more
likely, reused for multiple sessions. In this example, Alice calls
Bob. In this example we assume that Alice and Bob share the same
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proxy.
The example shows the SIP message flows where Alice acts as the
passive endpoint and Bob acts as the active endpoint meaning that as
soon as Bob receives the INVITE from Alice, with DTLS specified in
the 'm' line of the offer, Bob will begin to negotiate a DTLS
association with Alice for both RTP and RTCP streams. Early media
(RTP and RTCP) starts to flow from Bob to Alice as soon as Bob sends
the DTLS finished message to Alice. Bi-directional media (RTP and
RTCP) can flow after Bob sends the SIP 200 response and once Alice
has sent the DTLS finished message.
The SIP signaling from Alice to her proxy is transported over TLS to
ensure an integrity protected channel between Alice and her identity
service. Note that all other signaling is transported over TCP in
this example although it could be done over any supported transport.
Alice Proxies Bob
|(1) INVITE | |
|---------------->| |
| |(2) INVITE |
| |----------------->|
| | (3) hello |
|<-----------------------------------|
|(4) hello | |
|----------------------------------->|
| | (5) finished |
|<-----------------------------------|
| | (6) media |
|<-----------------------------------|
|(7) finished | |
|----------------------------------->|
| | (8) 200 OK |
|<-----------------------------------|
| | (9) media |
|----------------------------------->|
|(10) ACK | |
|----------------------------------->|
Message (1): INVITE Alice -> Proxy
This shows the initial INVITE from Alice to Bob carried over the
TLS transport protocol to ensure an integrity protected channel
between Alice and her proxy which acts as Alice's identity
service. Note that Alice has requested to be the passive endpoint
which means that it will act as the DTLS server and Bob will
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initiate the session. Also note that there is a fingerprint
attribute on the 'c' line of the SDP. This is computed from Bob's
self-signed certificate.
[[ NOTE: This example is not completely correct because the exact
syntax of the SDP is not yet determined. The MMUSIC working group
is currently working on standardizing mechanisms for SDP
capability negotiation which will enable this sort of best-effort
encryption. When that work is finished, this draft will be
harmonized with it.]]
INVITE sip:bob@example.com SIP/2.0
Via: SIP/2.0/TLS 192.168.1.101:5060;branch=z9hG4bK-0e53sadfkasldkfj
Max-Forwards: 70
Contact: <sip:alice@192.168.1.103:6937;transport=TLS>
To: <sip:bob@example.com>
From: "Alice"<sip:alice@example.com>;tag=843c7b0b
Call-ID: 6076913b1c39c212@REVMTEpG
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Type: application/sdp
Content-Length: xxxx
v=0
o=- 1181923068 1181923196 IN IP4 192.168.1.103
s=example1
c=IN IP4 192.168.1.103
a=setup:passive
a=connection:new
a=fingerprint: \
SHA-1 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
t=0 0
m=audio 6056 RTP/AVP 0
a=sendrecv
a=tcap:1 UDP/TLS/RTP/AVP RTP/AVP
a=pcfg:1 t=1
Message (2): INVITE Proxy -> Bob
This shows the INVITE being relayed to Bob from Alice (and Bob's)
proxy. Note that Alice's proxy has inserted an Identity and
Identity-Info header. This example only shows one element for
both proxies for the purposes of simplification. Bob verifies the
identity provided with the INVITE. Note that this offer includes
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a default m-line offering RTP in case the answerer does not
support SRTP. However, the potential configuration utilizing a
transport of SRTP is preferred. See
[I-D.ietf-mmusic-sdp-capability-negotiation] for more details on
the details of SDP capability negotiation.
INVITE sip:bob@example.com SIP/2.0
Via: SIP/2.0/TLS 192.168.1.101:5060;branch=z9hG4bK-0e53sadfkasldkfj
Via: SIP/2.0/TCP 192.168.1.100:5060;branch=z9hG4bK-0e53244234324234
Via: SIP/2.0/TCP 192.168.1.103:6937;branch=z9hG4bK-0e5b7d3edb2add32
Max-Forwards: 70
Contact: <sip:alice@192.168.1.103:6937;transport=TLS>
To: <sip:bob@example.com>
From: "Alice"<sip:alice@example.com>;tag=843c7b0b
Call-ID: 6076913b1c39c212@REVMTEpG
CSeq: 1 INVITE
Identity: CyI4+nAkHrH3ntmaxgr01TMxTmtjP7MASwliNRdupRI1vpkXRvZXx1ja9k
3W+v1PDsy32MaqZi0M5WfEkXxbgTnPYW0jIoK8HMyY1VT7egt0kk4XrKFC
HYWGCl0nB2sNsM9CG4hq+YJZTMaSROoMUBhikVIjnQ8ykeD6UXNOyfI=
Identity-Info: https://example.com/cert
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Type: application/sdp
Content-Length: xxxx
v=0
o=- 1181923068 1181923196 IN IP4 192.168.1.103
s=example1
c=IN IP4 192.168.1.103
a=setup:passive
a=connection:new
a=fingerprint: \
SHA-1 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
t=0 0
m=audio 6056 RTP/AVP 0
a=sendrecv
a=tcap:1 UDP/TLS/RTP/SAVP RTP/AVP
a=pcfg:1 t=1
Message (3): ClientHello Bob -> Alice
Assuming that Alice's identity is valid, Message 3 shows Bob
sending a DTLS ClientHello directly to Alice for each 'm' line in
the SDP. In this case two DTLS ClientHello messages are sent to
Alice. Bob sends a DTLS ClientHello to 192.168.1.103:6056 for RTP
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and another to port 6057 for RTCP.
Message (4): ServerHello+Certificate Alice -> Bob
Alice sends back a ServerHello, Certificate, ServerHelloDone for
both RTP and RTCP associations. Note that the same certificate is
used for both the RTP and RTCP associations. If RTP/RTCP
multiplexing [I-D.ietf-avt-rtp-and-rtcp-mux] were being used only
a single association would be required.
Message (5): Certificate Bob -> Alice
Bob sends a Certificate, ClientKeyExchange, CertificateVerify,
change_cipher_spec and Finished for both RTP and RTCP
associations. Again note that Bob uses the same server
certificate for both associations.
Message (6): Early Media Bob -> Alice
At this point, Bob can begin sending early media (RTP and RTCP) to
Alice. Note that Alice can't yet trust the media since the
fingerprint has not yet been received. This lack of trusted,
secure media is indicated to Alice.
Message (7): Finished Alice -> Bob
After Message 5 is received by Bob, Alice sends change_cipher_spec
and Finished.
Message (8): 200 OK Bob -> Alice
When Bob answers the call, Bob sends a 200 OK SIP message which
contains the fingerprint for Bob's certificate. When Alice
receives the message and validates the certificate presented in
Message 5. The endpoint now shows Alice that the call as secured.
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SIP/2.0 200 OK
To: <sip:bob@example.com>;tag=6418913922105372816
From: "Alice" <sip:alice@example.com>;tag=843c7b0b
Via: SIP/2.0/TCP 192.168.1.103:6937;branch=z9hG4bK-0e5b7d3edb2add32
Call-ID: 6076913b1c39c212@REVMTEpG
CSeq: 1 INVITE
Contact: <sip:192.168.1.104:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: xxxx
v=0
o=- 6418913922105372816 2105372818 IN IP4 192.168.1.104
s=example2
c=IN IP4 192.168.1.104
a=setup:active
a=connection:new
a=fingerprint:\
SHA-1 FF:FF:FF:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
t=0 0
m=audio 12000 UDP/TLS/RTP/SAVP 0
a=rtpmap:0 PCMU/8000/1
a=acfg:1 t=1
Message (9): RTP+RTCP Alice -> Bob
At this point, Alice can also start sending RTP and RTCP to Bob.
Note that in this case, Bob signals the actual transport protocol
configuration of SRTP over DTLS in the acfg parameter.
Message 10: ACK Alice -> Bob
Finally, Alice sends the SIP ACK to Bob.
8. Security Considerations
DTLS or TLS media signalled with SIP requires a way to ensure that
the communicating peers' certificates are correct.
The standard TLS/DTLS strategy for authenticating the communicating
parties is to give the server (and optionally the client) a PKIX
[RFC3280] certificate. The client then verifies the certificate and
checks that the name in the certificate matches the server's domain
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name. This works because there are a relatively small number of
servers with well-defined names; a situation which does not usually
occur in the VoIP context.
The design described in this document is intended to leverage the
authenticity of the signaling channel (while not requiring
confidentiality). As long as each side of the connection can verify
the integrity of the SDP INVITE then the DTLS handshake cannot be
hijacked via a man-in-the-middle attack. This integrity protection
is easily provided by the caller to the callee (see Alice to Bob in
Section 7) via the SIP Identity [RFC4474] mechanism. However, it is
less straightforward for the responder.
Ideally Alice would want to know that Bob's SDP had not been tampered
with and who it was from so that Alice's User Agent could indicate to
Alice that there was a secure phone call to Bob. This is known as the
SIP connected party problem and is still a topic of ongoing work in
the SIP community. In the meantime, there are several approaches
that can be used to mitigate this problem: Use UPDATE, Use SIPS, Use
S/MIME, Single Sided Verification, or use human-read Short
Authentication String (SAS) to validate the certificates. Each one
is discussed here followed by the security implications of that
approach.
8.1. UPDATE
[RFC4916] defines an approach for a UA to supply its identity to its
peer UA and for this identity to be signed by an authentication
service. For example, using this approach, Bob sends an answer, then
immediately follows up with an UPDATE that includes the fingerprint
and uses the SIP Identity mechanism to assert that the message is
from Bob@example.com. The downside of this approach is that it
requires the extra round trip of the UPDATE. However, it is simple
and secure even when not all of the proxies are trusted. In this
example, Bob only needs to trust his proxy.
[[OPEN ISSUE: Note that there is a window of vulnerability during
the early media phase of this operation before Alice receives the
UPDATE (which immediately follows the SDP answer). During this
window, Alice cannot be sure of Bob's identity. This risk might be
mitigated by including a secret in the offer which must be used to
establish the DTLS association, for instance via TLS PSK [RFC4279].
We are still studying this issue. Obviously, this is more attractive
if SIPS is used.]]
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8.2. SIPS
In this approach, the signaling is protected by TLS from hop to hop.
As long as all proxies are trusted, this provides integrity for the
fingerprint. It does not provide a strong assertion of who Alice is
communicating with. However, as much as the target domain can be
trusted to correctly populate the From header field value, Alice can
use that. The security issue with this approach is that if one of
the Proxies wished to mount a man-in-the-middle attack, it could
convince Alice that she was talking to Bob when really the media was
flowing through a man in the middle media relay. However, this
attack could not convince Bob that he was taking to Alice.
8.3. S/MIME
RFC 3261 [RFC3261] defines a S/MIME security mechanism for SIP that
could be used to sign that the fingerprint was from Bob. This would
be secure. However, so far there have been no deployments of S/MIME
for SIP.
8.4. Single-sided Verification
In this approach, no integrity is provided for the fingerprint from
Bob to Alice. In this approach, an attacker that was on the
signaling path could tamper with the fingerprint and insert
themselves as a man-in-the-middle on the media. Alice would know
that she had a secure call with someone but would not know if it was
with Bob or a man-in-the-middle. Bob would know that an attack was
happening. The fact that one side can detect this attack means that
in most cases where Alice and Bob both wish the communications to be
encrypted there is not a problem. Keep in mind that in any of the
possible approaches Bob could always reveal the media that was
received to anyone. We are making the assumption that Bob also wants
secure communications. In this do nothing case, Bob knows the media
has not been tampered with or intercepted by a third party and that
it is from Alice@example.com. Alice knows that she is talking to
someone and that whoever that is has probably checked that the media
is not being intercepted or tampered with. This approach is
certainly less than ideal but very usable for many situations.
8.5. Continuity of Authentication
One desirable property of a secure media system is to provide
continuity of authentication: being able to ensure cryptographically
that you are talking to the same person as before. With DTLS,
continuity of authentication is achieved by having each side use the
same public key/self-signed certificate for each connection (at least
with a given peer entity). It then becomes possible to cache the
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credential (or its hash) and verify that it is unchanged. Thus, once
a single secure connection has been established, an implementation
can establish a future secure channel even in the face of future
insecure signalling.
In order to enable continuity of authentication, implementations
SHOULD attempt to keep a constant long-term key. Verifying
implementations SHOULD maintain a cache of the key used for each peer
identity and alert the user if that key changes.
8.6. Short Authentication String
An alternative available to Alice and Bob is to use human speech to
verify each others' identity and then to verify each others'
fingerprints also using human speech. Assuming that it is difficult
to impersonate another's speech and seamlessly modify the audio
contents of a call, this approach is relatively safe. It would not
be effective if other forms of communication were being used such as
video or instant messaging. DTLS supports this mode of operation.
The minimal secure fingerprint length is around 64 bits.
ZRTP [I-D.zimmermann-avt-zrtp] includes Short Authentication String
mode in which a unique per-connection bitstring is generated as part
of the cryptographic handshake. The SAS can be as short as 25 bits
and so is somewhat easier to read. DTLS does not natively support
this mode, however it would be straightforward to add one as a TLS
extension [RFC3546].
8.7. Perfect Forward Secrecy
One concern about the use of a long-term key is that compromise of
that key may lead to compromise of past communications. In order to
prevent this attack, DTLS supports modes with Perfect Forward Secrecy
using Diffie-Hellman and Elliptic-Curve Diffie-Hellman cipher suites.
When these modes are in use, the system is secure against such
attacks. Note that compromise of a long-term key may still lead to
future active attacks. If this is a concern, a backup authentication
channel such as manual fingerprint establishment or a short
authentication string should be used.
9. IANA Considerations
This specification does not require any IANA actions.
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10. Acknowledgments
Cullen Jennings contributed substantial text and comments to this
document. This document benefited from discussions with Francois
Audet, Nagendra Modadugu, and Dan Wing. Thanks also for useful
comments by Flemming Andreasen, Rohan Mahy, David McGrew, Miguel
Garcia, Steffen Fries, Brian Stucker, and David Oran.
We would like to thank Thomas Belling, Guenther Horn, Steffen Fries,
Brian Stucker, Francois Audet, Dan Wing, Jari Arkko, and Vesa
Lehtovirta for their input regarding traversal of SBCs.
11. References
11.1. Normative References
[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.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3280] Housley, R., Polk, W., Ford, W., and D. Solo, "Internet
X.509 Public Key Infrastructure Certificate and
Certificate Revocation List (CRL) Profile", RFC 3280,
April 2002.
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private
Extensions to the Session Initiation Protocol (SIP) for
Asserted Identity within Trusted Networks", RFC 3325,
November 2002.
[RFC3489] Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
"STUN - Simple Traversal of User Datagram Protocol (UDP)
Through Network Address Translators (NATs)", RFC 3489,
March 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
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[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
the Session Description Protocol (SDP)", RFC 4145,
September 2005.
[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", RFC 4347, April 2006.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the
Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)", RFC 4572, July 2006.
[I-D.fischl-mmusic-sdp-dtls]
Fischl, J. and H. Tschofenig, "Session Description
Protocol (SDP) Indicators for Datagram Transport Layer
Security (DTLS)", draft-fischl-mmusic-sdp-dtls-03 (work in
progress), July 2007.
11.2. Informational References
[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
and RTP Control Protocol (RTCP) Packets over Connection-
Oriented Transport", RFC 4571, July 2006.
[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.
[RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
Carrara, "Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming
Protocol (RTSP)", RFC 4567, July 2006.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[RFC4346] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.1", RFC 4346, April 2006.
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[I-D.zimmermann-avt-zrtp]
Zimmermann, P., "ZRTP: Media Path Key Agreement for Secure
RTP", draft-zimmermann-avt-zrtp-04 (work in progress),
July 2007.
[I-D.mcgrew-srtp-ekt]
McGrew, D., "Encrypted Key Transport for Secure RTP",
draft-mcgrew-srtp-ekt-03 (work in progress), July 2007.
[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-00 (work in progress), July 2007.
[I-D.ietf-sip-media-security-requirements]
Wing, D., Fries, S., Tschofenig, H., and F. Audet,
"Requirements and Analysis of Media Security Key
Management Protocols",
draft-ietf-sip-media-security-requirements-00 (work in
progress), September 2007.
[I-D.ietf-mmusic-sdp-capability-negotiation]
Andreasen, F., "SDP Capability Negotiation",
draft-ietf-mmusic-sdp-capability-negotiation-07 (work in
progress), October 2007.
[I-D.ietf-avt-rtp-and-rtcp-mux]
Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port",
draft-ietf-avt-rtp-and-rtcp-mux-07 (work in progress),
August 2007.
[RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of
Provisional Responses in Session Initiation Protocol
(SIP)", RFC 3262, June 2002.
[RFC3546] Blake-Wilson, S., Nystrom, M., Hopwood, D., Mikkelsen, J.,
and T. Wright, "Transport Layer Security (TLS)
Extensions", RFC 3546, June 2003.
[RFC4916] Elwell, J., "Connected Identity in the Session Initiation
Protocol (SIP)", RFC 4916, June 2007.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
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[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
[RFC4279] Eronen, P. and H. Tschofenig, "Pre-Shared Key Ciphersuites
for Transport Layer Security (TLS)", RFC 4279,
December 2005.
[I-D.wing-sipping-srtp-key]
Wing, D., "Disclosing Secure RTP (SRTP) Session Keys with
a SIP Event Package", draft-wing-sipping-srtp-key-01 (work
in progress), July 2007.
Appendix A. Requirements Analysis
[I-D.ietf-sip-media-security-requirements] describes security
requirements for media keying. This section evaluates this proposal
with respect to each requirement.
A.1. Forking and retargeting (R1, R2, R3)
In this draft, the SDP offer (in the INVITE) is simply an
advertisement of the capability to do security. This advertisement
does not depend on the identity of the communicating peer, so forking
and retargeting work work when all the endpoints will do SRTP. When
a mix of SRTP and non-SRTP endpoints are present, we expect to use
the SDP capabilities mechanism currently being defined
[I-D.ietf-mmusic-sdp-capability-negotiation] to transparently
negotiate security where possible. Because DTLS establishes a new
key for each session, only the entity with which the call is finally
established gets the media encryption keys (R3).
A.2. Reusage of a Security Context (R4), (R11)
DTLS allows sessions to be resumed with the 'TLS session resumption'
functionality. This feature can be used to lower the amount of
cryptographic computation that needs to be done when two peers re-
initiates the communication.
A.3. Clipping (R5)
Because the key establishment occurs in the media plane, media need
not be clipped before the receipt of the SDP answer.
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A.4. Passive Attacks on the Media Path (R6)
The public key algorithms used by DTLS ciphersuites, such as RSA,
Diffie-Hellman, and Elliptic Curve Diffie-Hellman, are secure against
passive attacks.
A.5. Passive Attacks on the Signaling Path (R7)
DTLS provides protection against passive attacks by adversaries on
the signaling path since only a fingerprint is exchanged using SIP
signaling.
A.6. Perfect Forward Secrecy (R8)
DTLS supports Diffie-Hellman and Elliptic Curve Diffie-Hellman cipher
suites which provide PFS.
A.7. Algorithm Negotiation (R9)
DTLS negotiates cipher suites before performing significant
cryptographic computation and therefore supports algorithm
negotiation and multiple cipher suites without additional
computational expense.
A.8. RTP Validity Check (R10)
TBD
A.9. 3rd Party Certificates (R12, R18)
Third party certificates are not required. However, if the parties
share an authentication infrastructure that is compatible with TLS
(3rd party certificates or shared keys) it can be used.
A.10. FIPS 140-2 (R13)
TLS implementations already may be FIPS 140-2 approved and the
algorithms used here are consistent with the approval of DTLS and
DTLS-SRTP.
A.11. Linkage between Keying Exchange and SIP Signaling (R14)
The signaling exchange is linked to the key management exchange using
the fingerprints carried in SIP and the certificates are exchanged in
DTLS.
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A.12. Start with RTP and Upgrade to SRTP (R15)
DTLS-SRTP as described in this framework does not require an SRTP
security context to be established as part of the initial
communication setup. Instead, the DTLS handshake can be initiated
later during on ongoing session.
A.13. Denial of Service Vulnerability (R16)
DTLS offers some degree of DoS protection particuarly as a built-in
feature.
A.14. Adversary Model (R17)
DTLS-SRTP requires that an adversary is at least able to intercept
the fingerprint exchange along the SIP signaling path (i.e., active
attack) and to intercept the DTLS handshake by acting as a man-in-
the-middle adversary (i.e., active attack).
A.15. Crypto-Agility (R19)
DTLS allows ciphersuites to be negotiated and hence new algorithms
can be incrementally deployed. Work on replacing the fixed MD5/SHA-1
key derivation function is ongoing.
A.16. Downgrading Protection (R20)
DTLS provides protection against downgrading attacks since the
selection of the offered ciphersuites is confirmed in a later stage
of the handshake. This protection is efficient unless an adversary
is able to break a ciphersuite in real-time.
A.17. Media Security Negotation (R21)
DTLS allows a User Agent to negotiate media security parameters for
each individual session.
A.18. Signaling Protocol Independence (R22)
The DTLS-SRTP framework does not rely on SIP; every protocol that is
capable of exchanging a fingerprint and the media description can be
secured.
A.19. Media Recording (R23)
An extension, see [I-D.wing-sipping-srtp-key], has been specified to
support media recording that does not require intermediaries to act
as a MITM.
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When media recording is done by intermediaries then they need to act
as a MITM.
A.20. Lawful Interception (R24)
Lawful interception requires an active MITM who is located along the
signaling and the data path.
A.21. Interworking with Intermediaries (R25)
A description of the interworking with Session Border Controllers is
described in this document.
A.22. PSTN Gateway Termination (R26)
The DTLS-SRTP framework allows the media security to terminate at a
PSTN gateway. [Editor's Note: A detailed description will be
provided in a future version of this document.]
Authors' Addresses
Jason Fischl
CounterPath Solutions, Inc.
Suite 300, One Bentall Centre, 505 Burrard Street
Vancouver, BC V7X 1M3
Canada
Phone: +1 604 320-3340
Email: jason@counterpath.com
Hannes Tschofenig
Nokia Siemens Networks
Otto-Hahn-Ring 6
Munich, Bavaria 81739
Germany
Email: Hannes.Tschofenig@nsn.com
URI: http://www.tschofenig.com
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Eric Rescorla
Network Resonance
2483 E. Bayshore #212
Palo Alto, CA 94303
USA
Email: ekr@networkresonance.com
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Fischl, et al. Expires May 15, 2008 [Page 28]
