SIPPING J. Fischl
Internet-Draft CounterPath Solutions, Inc.
Expires: December 27, 2006 H. Tschofenig
E. Rescorla
Network Resonance
June 25, 2006
Datagram Transport Layer Security (DTLS) Protocol for Protection of
Media Traffic Established with the Session Initiation Protocol
draft-fischl-sipping-media-dtls-01.txt
Status of this Memo
By submitting this Internet-Draft, each author represents that any
applicable patent or other IPR claims of which he or she is aware
have been or will be disclosed, and any of which he or she becomes
aware will be disclosed, in accordance with Section 6 of BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt.
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
This Internet-Draft will expire on December 27, 2006.
Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
This document specifies how to use the Session Initiation Protocol
(SIP) to establish secure media sessions using or over the Datagram
Transport Layer Security (DTLS) protocol. It describes a mechanism
of transporting a fingerprint attribute in the Session Description
Fischl, et al. Expires December 27, 2006 [Page 1]
Internet-Draft SIP for DTLS Media Traffic June 2006
Protocol (SDP) that identifies the certificate that will be presented
during the DTLS handshake. It relies on the SIP identity mechanism
to ensure the integrity of the fingerprint attribute. This allows
the establishment of media security along the media path.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Motivation . . . . . . . . . . . . . . . . . . . . . . . . . . 5
4. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5
5. Verifying Certificate Integrity . . . . . . . . . . . . . . . 6
6. Miscellaneous Considerations . . . . . . . . . . . . . . . . . 7
6.1. Anonymous Calls . . . . . . . . . . . . . . . . . . . . . 7
6.2. Early Media . . . . . . . . . . . . . . . . . . . . . . . 8
6.3. Forking . . . . . . . . . . . . . . . . . . . . . . . . . 8
6.4. Delayed Offer Calls . . . . . . . . . . . . . . . . . . . 8
6.5. Session Modification . . . . . . . . . . . . . . . . . . . 8
6.6. UDP Payload De-multiplex . . . . . . . . . . . . . . . . . 9
6.7. Rekeying . . . . . . . . . . . . . . . . . . . . . . . . . 9
6.8. Conference Servers and Shared Encryptions Contexts . . . . 9
6.9. Media over SRTP . . . . . . . . . . . . . . . . . . . . . 10
7. Example Message Flow . . . . . . . . . . . . . . . . . . . . . 10
8. Security Considerations . . . . . . . . . . . . . . . . . . . 15
8.1. UPDATE . . . . . . . . . . . . . . . . . . . . . . . . . . 16
8.2. SIPS . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
8.3. S/MIME . . . . . . . . . . . . . . . . . . . . . . . . . . 16
8.4. Single-sided Verification . . . . . . . . . . . . . . . . 17
8.5. Short Authentication String . . . . . . . . . . . . . . . 17
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 18
10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 18
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
11.1. Normative References . . . . . . . . . . . . . . . . . . . 18
11.2. Informational References . . . . . . . . . . . . . . . . . 19
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22
Intellectual Property and Copyright Statements . . . . . . . . . . 23
Fischl, et al. Expires December 27, 2006 [Page 2]
Internet-Draft SIP for DTLS Media Traffic June 2006
1. Introduction
The Session Initiation Protocol (SIP) RFC 3261 [RFC3261] and the
Session Description Protocol (SDP) [I-D.ietf-mmusic-sdp-new] are used
to set up multimedia sessions or calls. SDP is also used to set up
TCP [I-D.ietf-mmusic-sdp-comedia] and additionally TCP/TLS
connections for usage with media sessions [I-D.ietf-mmusic-comedia-
tls]. The Real-Time Protocol (RTP) RFC 3550 [RFC3550] is used to
transmit real time media on top of UDP, TCP [I-D.ietf-avt-rtp-
framing-contrans], and TLS [I-D.ietf-mmusic-comedia-tls]. Datagram
TLS RFC 4347 [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.mcgrew-tls-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 use self-signed certificates. The certificate
fingerprints are sent in SDP over SIP as part of the offer/answer
exchange. The SIP Identity mechanism [I-D.ietf-sip-identity] is used
to provide integrity for the fingerprints.
This approach differs from previous attempts to secure media traffic
where the authentication and key exchange protocol (e.g., MIKEY RFC
3830 [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
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.
Fischl, et al. Expires December 27, 2006 [Page 3]
Internet-Draft SIP for DTLS Media Traffic June 2006
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
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
Fischl, et al. Expires December 27, 2006 [Page 4]
Internet-Draft SIP for DTLS Media Traffic June 2006
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 [I-D.ietf-
sip-identity]. 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 [I-D.ietf-mmusic-sdescriptions], Key Management
Extensions [I-D.ietf-mmusic-kmgmt-ext] combined with MIKEY RFC 3830
[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 RFC 3262 [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-RTP [I-D.mcgrew-tls-
srtp] is used.
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 RFC 2119 [RFC2119].
Fischl, et al. Expires December 27, 2006 [Page 5]
Internet-Draft SIP for DTLS Media Traffic June 2006
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 RFC 3264 [RFC3264], is used by
protocols like the Session Initiation Protocol (SIP) RFC 3261
[RFC3261] to set up multimedia sessions. In addition to the usual
contents of an SDP [I-D.ietf-mmusic-sdp-new] message, each 'm' line
will also contain several attributes as specified in [I-D.fischl-
mmusic-sdp-dtls], RFC 4145 [RFC4145] and [I-D.ietf-mmusic-comedia-
tls].
The endpoint MUST use the setup and connection attributes defined in
RFC 4145 [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 [I-D.ietf-mmusic-comedia-tls].
The setup:active endpoint establishes a DTLS association with the
setup:passive endpoint RFC 4145 [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.
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
Fischl, et al. Expires December 27, 2006 [Page 6]
Internet-Draft SIP for DTLS Media Traffic June 2006
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 [I-D.ietf-sip-identity].
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.
Additionally, it MUST be ensured that the Privacy header RFC 3325
[RFC3325] is used in conjunction with the SIP identity mechanism to
ensure that the identity of the user is not asserted when enabling
Fischl, et al. Expires December 27, 2006 [Page 7]
Internet-Draft SIP for DTLS Media Traffic June 2006
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 RFC 3262 [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.
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
Fischl, et al. Expires December 27, 2006 [Page 8]
Internet-Draft SIP for DTLS Media Traffic June 2006
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 RFC
4145 [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 RFC 3489
[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 [I-D.ietf-tls-
rfc2246-bis]. 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.mcgrew-tls-srtp].
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
Fischl, et al. Expires December 27, 2006 [Page 9]
Internet-Draft SIP for DTLS Media Traffic June 2006
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.
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.mcgrew-tls-
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.mcgrew-tls-srtp].
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, if
desired, reused for multiple sessions. In this example, Alice calls
Bob. In this example we assume that Alice and Bob share the same
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.
Fischl, et al. Expires December 27, 2006 [Page 10]
Internet-Draft SIP for DTLS Media Traffic June 2006
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
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.
Fischl, et al. Expires December 27, 2006 [Page 11]
Internet-Draft SIP for DTLS Media Traffic June 2006
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 UDP/TLS/RTP/AVP 0
a=sendrecv
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.
Fischl, et al. Expires December 27, 2006 [Page 12]
Internet-Draft SIP for DTLS Media Traffic June 2006
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 UDP/TLS/RTP/AVP 0
a=sendrecv
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
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.
Fischl, et al. Expires December 27, 2006 [Page 13]
Internet-Draft SIP for DTLS Media Traffic June 2006
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.
Fischl, et al. Expires December 27, 2006 [Page 14]
Internet-Draft SIP for DTLS Media Traffic June 2006
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/AVP 0
a=rtpmap:0 PCMU/8000/1
Message (9): RTP+RTCP Alice -> Bob
At this point, Alice can also start sending RTP and RTCP to Bob
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 RFC
3280 [RFC3280] certificate. The client then verifies the certificate
and checks that the name in the certificate matches the server's
domain 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.
Fischl, et al. Expires December 27, 2006 [Page 15]
Internet-Draft SIP for DTLS Media Traffic June 2006
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 [I-D.ietf-sip-identity] 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
[I-D.ietf-sip-connected-identity] 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.
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
Fischl, et al. Expires December 27, 2006 [Page 16]
Internet-Draft SIP for DTLS Media Traffic June 2006
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. 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 can also be used with a Short Authentication String
authentication mechanism like that used by ZRTP [I-D.zimmermann-avt-
zrtp]. The DTLS short authentication string is computed as:
sasvalue=SHA-1(client_cert || ClientRandom
|| server_cert || ServerRandom)
where client_cert and server_cert are the DER encodings of the end-
entity certificates for client and server and ClientRandom and
ServerRandom are the random values from the handshake. The low-order
25 bits of this value should be used as the SAS and then represented
in base32 encoding RFC 3548 [RFC3548] and displayed to the users so
they can be verified.
Fischl, et al. Expires December 27, 2006 [Page 17]
Internet-Draft SIP for DTLS Media Traffic June 2006
9. IANA Considerations
This specification does not require any IANA actions.
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, and David
Oran.
11. References
11.1. Normative References
[I-D.ietf-mmusic-comedia-tls]
Lennox, J., "Connection-Oriented Media Transport over the
Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)",
draft-ietf-mmusic-comedia-tls-06 (work in progress),
March 2006.
[I-D.ietf-mmusic-sdp-new]
Handley, M., "SDP: Session Description Protocol",
draft-ietf-mmusic-sdp-new-26 (work in progress),
January 2006.
[I-D.ietf-sip-identity]
Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", draft-ietf-sip-identity-06
(work in progress), October 2005.
[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.
Fischl, et al. Expires December 27, 2006 [Page 18]
Internet-Draft SIP for DTLS Media Traffic June 2006
[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.
[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.
[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-00 (work in
progress), March 2006.
11.2. Informational References
[I-D.ietf-avt-rtp-framing-contrans]
Lazzaro, J., "Framing RTP and RTCP Packets over
Connection-Oriented Transport",
draft-ietf-avt-rtp-framing-contrans-06 (work in progress),
September 2005.
[I-D.ietf-mmusic-ice]
Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Methodology for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols",
draft-ietf-mmusic-ice-08 (work in progress), March 2006.
[I-D.ietf-mmusic-kmgmt-ext]
Arkko, J., "Key Management Extensions for Session
Fischl, et al. Expires December 27, 2006 [Page 19]
Internet-Draft SIP for DTLS Media Traffic June 2006
Description Protocol (SDP) and Real Time Streaming
Protocol (RTSP)", draft-ietf-mmusic-kmgmt-ext-15 (work in
progress), June 2005.
[I-D.ietf-mmusic-sdescriptions]
Andreasen, F., "Session Description Protocol Security
Descriptions for Media Streams",
draft-ietf-mmusic-sdescriptions-12 (work in progress),
September 2005.
[I-D.ietf-mmusic-sdp-comedia]
Yon, D., "Connection-Oriented Media Transport in the
Session Description Protocol (SDP)",
draft-ietf-mmusic-sdp-comedia-10 (work in progress),
November 2004.
[I-D.ietf-tls-rfc2246-bis]
Dierks, T. and E. Rescorla, "The TLS Protocol Version
1.1", draft-ietf-tls-rfc2246-bis-13 (work in progress),
June 2005.
[I-D.ietf-sip-connected-identity]
Elwell, J., "Connected Identity in the Session Initiation
Protocol (SIP)", draft-ietf-sip-connected-identity-00
(work in progress), April 2006.
[I-D.zimmermann-avt-zrtp]
Zimmermann, P., "ZRTP: Extensions to RTP for Diffie-
Hellman Key Agreement for SRTP",
draft-zimmermann-avt-zrtp-01 (work in progress),
March 2006.
[I-D.mcgrew-srtp-ekt]
McGrew, D., "Encrypted Key Transport for Secure RTP",
draft-mcgrew-srtp-ekt-00 (work in progress),
February 2006.
[I-D.mcgrew-tls-srtp]
Rescorla, E. and D. McGrew, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for Secure
Real-time Transport Protocol (SRTP)",
draft-mcgrew-tls-srtp-00 (work in progress), June 2006.
[RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of
Provisional Responses in Session Initiation Protocol
(SIP)", RFC 3262, June 2002.
[RFC3548] Josefsson, S., "The Base16, Base32, and Base64 Data
Fischl, et al. Expires December 27, 2006 [Page 20]
Internet-Draft SIP for DTLS Media Traffic June 2006
Encodings", RFC 3548, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
Fischl, et al. Expires December 27, 2006 [Page 21]
Internet-Draft SIP for DTLS Media Traffic June 2006
Authors' Addresses
Jason Fischl
CounterPath Solutions, Inc.
8th Floor, 100 West Pender Street
Vancouver, BC V6B 1R8
Canada
Phone: +1 604 320-3340
Email: jason@counterpath.com
Hannes Tschofenig
Email: Hannes.Tschofenig@gmx.net
Eric Rescorla
Network Resonance
2483 E. Bayshore #212
Palo Alto, CA 94303
USA
Email: ekr@networkresonance.com
Fischl, et al. Expires December 27, 2006 [Page 22]
Internet-Draft SIP for DTLS Media Traffic June 2006
Intellectual Property Statement
The IETF takes no position regarding the validity or scope of any
Intellectual Property Rights or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; nor does it represent that it has
made any independent effort to identify any such rights. Information
on the procedures with respect to rights in RFC documents can be
found in BCP 78 and BCP 79.
Copies of IPR disclosures made to the IETF Secretariat and any
assurances of licenses to be made available, or the result of an
attempt made to obtain a general license or permission for the use of
such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository at
http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at
ietf-ipr@ietf.org.
Disclaimer of Validity
This document and the information contained herein are provided on an
"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET
ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Copyright Statement
Copyright (C) The Internet Society (2006). This document is subject
to the rights, licenses and restrictions contained in BCP 78, and
except as set forth therein, the authors retain all their rights.
Acknowledgment
Funding for the RFC Editor function is currently provided by the
Internet Society.
Fischl, et al. Expires December 27, 2006 [Page 23]