SIP/2.0 487 Request Terminated
To: <sip:bob@example.org>;tag=6472
From: <sip:alice@example.org>;tag=7743
Call-ID: 425928@phone.example.org
CSeq: 1 INVITE
8. Security Considerations
The extension specified in this document significantly changes the
relative security of SIP devices. Currently in SIP, even if an
eavesdropper learns the Call-ID, To, and From headers of a dialog,
they cannot easily modify or destroy that dialog if Digest
authentication or end-to-end message integrity are used.
This extension can be used to disconnect participants or replace
participants in a multimedia conversation. As such, invitations with
the Replaces header MUST only be accepted if the peer requesting
replacement has been properly authenticated using a standard SIP
mechanism (Digest or S/MIME), and authorized to request a replacement
of the target dialog. All SIP implementations are already required
to support Digest Authentication. In addition, implementations which
support the Replaces header SHOULD also implement the Referred-By
mechanism.
How a User Agent determines which requests are legitimately
authorized to make dialog replacements is non-trivial and depends on
a considerable amount of local policy configuration. In general,
there are four cases when an authorization for a replacement is
reasonable or warranted.
1. Replacement made by a party considered equivalent to the replaced
party
2. Replacement made on behalf of the replaced party (perhaps
transitively)
3. Replacement made by a former participant
4. Replacement made by a specifically authorized party
Starting with #1 for example, if an executive and an assistant both
receive requests for a shared address-of-record, if so configured,
either should be able to replace dialogs of the other for the shared
identity. Both could even share the same keying material (Digest or
S/MIME), or one could hold an authorization document signed by the
other expressing this relationship. Likewise, in a call center
environment, each call center agent could possess credentials to
which supervisors also have access.
The most common use case of a replacement is on the request of the
replaced participant (who no longer wants to be involved). This is
the case in many features, such as completing an Attended Transfer
and converting a 3-way call to a point-to-point call. Such
replacements are typically triggered by a REFER [8] request from the
replaced participant. The Referred-By [4] mechanism defines one way
to identify the apparent original requester and can point to a SIP
Authenticated Identity Body [5] (an S/MIME-based signed assertion) to
secure this information.
In the example in section 1, Alice sends an INVITE with Replaces to
Bob. Alice was a former participant in the conversation and had a
previous dialog relationship with Bob. Alice can use the same Digest
or S/MIME credentials she used to authenticate with Bob during the
original call to prove that she was a former participant. Note that
this justification for replacing calls is more dangerous than the
others, and in most cases is another way to authorize that the
replacing participant is available. Implementations SHOULD NOT rely
on this method as an authorization mechanism.
The last scenario is the easiest to secure but the least likely to be
useful in practice. It is unlikely that an arbitrary host in the
Internet is aware of any special authorization relationship between
the replaced and the replacing parties. However, this use case may
be useful in some environments. Since this usage does not
effectively degrade the security of the solution, it is still
allowed.
Some mechanisms for obtaining the dialog information needed by the
Replaces header (Call-ID, to-tag, and from-tag) include URIs on a web
page, subscriptions to an appropriate event package, and
notifications after a REFER request. Since manipulating this dialog
information could cause User Agents to replace the wrong dialog, use
of message integrity protection for this information is STRONGLY
RECOMMENDED. Use of end-to-end security mechanisms to encrypt this
information is also RECOMMENDED.
This extension was designed to take advantage of future signature or
authorization schemes defined in standards track extensions. In
general, call control features benefit considerably from such work.
9. IANA Considerations
9.1. Registration of "Replaces" SIP header
Name of Header: Replaces
Short form: none
Normative description: section 6.1 of this document
9.2. Registration of "replaces" SIP Option-tag
Name of option: replaces
Description: Support for the SIP Replaces header
SIP headers defined: Replaces
Normative description: This document
10. Acknowledgments
Thanks to Robert Sparks, Alan Johnston, Dan Petrie, Ben Campbell, and
many other members of the SIP WG for their continued support of the
cause of distributed call control in SIP.
11. References
11.1. Normative References
[1] 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.
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[3] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 2234, November 1997.
[4] Sparks, R., "The Session Initiation Protocol (SIP) Referred-By
Mechanism", RFC 3892, September 2004.
[5] Peterson, J., "The Session Initiation Protocol (SIP)
Authenticated Identity Body (AIB) Format", RFC 3893, September
2004.
[6] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A., and L. Stewart, "HTTP Authentication:
Basic and Digest Access Authentication", RFC 2617, June 1999.
[7] Ramsdell, B., "Secure/Multipurpose Internet Mail Extensions
(S/MIME) Version 3.1 Message Specification", RFC 3851, July
2004.
11.2. Informative References
[8] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
[9] Handley, M., Schulzrinne, H., Schooler, E., and J. Rosenberg,
"SIP: Session Initiation Protocol", RFC 2543, March 1999.
[10] Mahy, R., "A Call Control and Multi-party usage framework for
the Session Initiation Protocol (SIP)", Work in Progress, March
2003.
[11] Rosenberg, J., Peterson, J., Schulzrinne, H., and G. Camarillo,
"Best Current Practices for Third Party Call Control (3pcc) in
the Session Initiation Protocol (SIP)", BCP 85, RFC 3725, April
2004.
[12] Sparks, R. and A. Johnston, "Session Initiation Protocol Call
Control - Transfer", Work in Progress, February 2003.
[13] Rosenberg, J. and H. Schulzrinne, "An INVITE Initiated Dialog
Event Package for the Session Initiation Protocol (SIP)", Work
in Progress, March 2003.
[14] Johnston, A. and S. Donovan, "Session Initiation Protocol
Service Examples", Work in Progress, March 2003.
[15] Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C., and
D. Gurle, "Session Initiation Protocol (SIP) Extension for
Instant Messaging", RFC 3428, December 2002.
[16] Roach, A., "Session Initiation Protocol (SIP)-Specific Event
Notification", RFC 3265, June 2002.
[17] Donovan, S., "The SIP INFO Method", RFC 2976, October 2000.
[18] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
Method", RFC 3311, October 2002.
[19] Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional
Responses in Session Initiation Protocol (SIP)", RFC 3262, June
2002.
[20] Campbell, B., "SIMPLE Presence Publication Mechanism", Work in
Progress, February 2003.
12. Authors’ Addresses
Rohan Mahy
Cisco Systems, Inc.
5617 Scotts Valley Dr
Scotts Valley, CA 95066
USA
EMail: rohan@cisco.com
Billy Biggs
EMail: bbiggs@dumbterm.net
Rick Dean
EMail: rfc@fdd.com
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