SIP C. Jennings
Internet-Draft Cisco Systems
Expires: January 13, 2006 N. Modadugu
Stanford University
July 12, 2005
Using DTLS as a Transport for SIP
draft-jennings-sip-dtls-01
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Copyright (C) The Internet Society (2005).
Abstract
This draft specifies how to use Datagram Transport Layer Security
(DTLS) as a transport for SIP. DTLS is a new protocol for providing
TLS security over a datagram protocol.
This draft is being discussed on the sip@ietf.org mailing list.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Transport Parameters . . . . . . . . . . . . . . . . . . . . . 3
4. DTLS Usage . . . . . . . . . . . . . . . . . . . . . . . . . . 4
5. Locating DTLS SIP Servers . . . . . . . . . . . . . . . . . . 4
6. Security Considerations . . . . . . . . . . . . . . . . . . . 5
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 5
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 5
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 5
9.1 Normative References . . . . . . . . . . . . . . . . . . . 5
9.2 Informational References . . . . . . . . . . . . . . . . . 6
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 6
Intellectual Property and Copyright Statements . . . . . . . . 7
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1. Introduction
Datagram Transport Layer Security (DTLS) [7] provides communication
privacy similar to TLS for datagram packets. SIP can run over both
stream and datagram transports, including UDP and TCP. SIP already
defines how to use TLS with stream oriented transports. This
specification extends SIP to use DTLS with datagram oriented
transports.
There has been considerable discussion of why SIP needs DTLS when we
have TLS. This is the wrong question. The right question is why SIP
has UDP and TCP (not to mention SCTP). There are two reasons for
believing that UDP is likely to be an important protocol in SIP for
the foreseeable future.
o In theory, there is no problem building systems that terminate a
million TCP connections on a single host. In practice, the common
operating systems used for building SIP aggregation devices make
this impossible. To date, no one has demonstrated terminating
over 100k SIP TCP connections to a single host. Doing that many
connections with UDP has not been difficult.
o If we want to talk about "running code" for SIP, it's UDP. Unless
UDP is deprecated for SIP, it is important to provide a reasonable
level of security for it.
The dominant technique for providing security for SIP/TCP is DTLS.
Although in principle S/MIME can be used to provide security for both
SIP/UDP and SIP/TCP, deployment has been minimal. Given that SIP/UDP
is not likely to disappear any time soon, this leaves us with two
options:
1. Leave UDP users without a channel-level security mechanism; or
2. Provide SIP/UDP users with an alternate security mechanism.
We don't consider the first option to be acceptable. The obvious
approach to the second option is to use DTLS, which provides UDP with
a level of channel security equivalent to that which TLS provides for
TCP.
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 RFC 2119 [3].
3. Transport Parameters
SIP URIs can carry a transport parameter indicating the transport
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protocol to be used. This specification defines two new values for
the transport parameter: "dtls-udp" for the SIP URI transport
parameter to be used for messages sent using DTLS over UDP, and
"dtls-dccp" for messages sent using DTLS over DCCP. The update to
the ABNF in RFC 3261 for this parameter is the following:
transport-param = "transport="
( "udp" / "tcp" / "sctp" / "tls" / "tls-sctp"
"dtls-dccp" / "dtls-udp"
/ other-transport)
The following is an example of SIP URIs using "dtls-udp":
sip:alice@example.com;transport=dtls-udp
Via header fields also carry a transport protocol identifier. This
specification extends RFC 3261 to define the value "DTLS-UDP" for
DTLS over UDP and "DTLS-DCCP" for DTLS over DCCP. The update to the
ABNF in RFC 3261 for this parameter is the following:
transport = "UDP" / "TCP" / "TLS" / "SCTP" / "TLS-SCTP"
"DTLS-DCCP" / "DTLS-UDP"
/ other-transport
The following is an example Via header field:
Via: SIP/2.0/DTLS-UDP atlanta.example.com:5060
4. DTLS Usage
The normal rules for sending a request over UDP in RFC 3261 apply to
sending over DTLS. Note that the congestion safety rules for UDP do
not apply to DCCP. In addition, the normal rules for validating a
TLS connection in RFC 3261 apply to DTLS connections. Requests with
a SIPS URI can be sent over DTLS as well as TLS.
5. Locating DTLS SIP Servers
The normal rules from RFC 3263 [4] apply when locating a SIP server
that supports DTLS. The following new NAPTR[5] service values are
defined: "SIPS+D2U" for UDP, and "SIPS+D2D" for DCCP. In addition,
the service value "SIP+D2D" should be used for SIP without DTLS over
DCCP.
The default port for DTLS over UDP is 5061.
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6. Security Considerations
The security issues with SIP using DTLS are equivalent to the issues
of using SIP with TLS. All the security considerations in RFC 3261
relevant to TLS apply to DTLS.
7. IANA Considerations
The IANA is requested to update the following entry to the "SIP/SIPS
URI Parameters" registry. The reference to this RFC should appear in
double-brackets and be appended to the list of references already
listed on for the transport parameter, as indicated in RFC 3969 [6].
The result is shown below:
Parameter Name Predefined Values Reference
-------------- ----------------- ---------
transport Yes [RFC3261] [[RFCXXXX]]
This document also defines new NAPTR service field values. The IANA
is requested to register these values under the "Registry for the SIP
SRV Resource Record Services Field". The resulting entries should
be:
Services Field Protocol Reference
-------------------- -------- ---------
SIPS+D2U UDP [RFCXXXX]
SIPS+D2D DCCP [RFCXXXX]
SIP+D2D DCCP [RFCXXXX]
[Note to IANA: Please replace XXXX with the RFC number of this
specification.]
8. Acknowledgments
Much of text and outline for this specification came from [8]
authored by Jonathan Rosenberg, Henning Schulzrinne, and Gonzalo
Camarillo. Eric Rescorla provided helpful comments and text.
9. References
9.1 Normative References
[1] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", draft-rescorla-dtls-05 (work in progress), June 2005.
[2] 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.
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[3] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[4] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
(SIP): Locating SIP Servers", RFC 3263, June 2002.
[5] Mealling, M., "Dynamic Delegation Discovery System (DDDS) Part
Three: The Domain Name System (DNS) Database", RFC 3403,
October 2002.
9.2 Informational References
[6] Camarillo, G., "The Internet Assigned Number Authority (IANA)
Uniform Resource Identifier (URI) Parameter Registry for the
Session Initiation Protocol (SIP)", BCP 99, RFC 3969,
December 2004.
[7] Kohler, E., "Datagram Congestion Control Protocol (DCCP)",
draft-ietf-dccp-spec-09 (work in progress), November 2004.
[8] Rosenberg, J., "The Stream Control Transmission Protocol (SCTP)
as a Transport for the Session Initiation Protocol (SIP)",
draft-ietf-sip-sctp-06 (work in progress), February 2005.
Authors' Addresses
Cullen Jennings
Cisco Systems
170 West Tasman Drive
MS: SJC-21/2
San Jose, CA 95134
USA
Phone: +1 408 902-3341
Email: fluffy@cisco.com
Nagendra Modadugu
Stanford University
353 Serra Mall
Stanford, CA 94305
USA
Email: Nagendra@cs.stanford.edu
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