6.3. Alice Calls Bob's SIP AOR Using TCP
Bob's registration has already occurred as per Section 6.1.
In the second example, Alice calls Bob's SIP AOR instead
(sip:bob@example.com), and she uses TCP as a transport. Registrar/
Authoritative Proxy B consults the binding in the registration
database, and finds the two Contact header field bindings. Alice had
addressed Bob with a SIP Request-URI (sip:bob@example.com), so
Registrar/Authoritative Proxy B determines that the call needs to be
routed both to bobpc (which registered with a SIP Contact header
field) and bobphone (which registered with a SIPS Contact header
field), and therefore the request is forked to
sip:bob@bobpc.example.com and sip:bob@bobphone.example.com, through
Edge Proxy B. Note that Registrar/Authoritative Proxy B preserved
the SIP scheme of the Request-URI instead of replacing it with the
SIPS scheme of the Contact header field that was used for
registration. Both Registrar/Authoritative Proxy B and Edge Proxy B
insert themselves in the Record-Route. Bob's phone's policy is to
accept calls to SIP and SIPS (i.e., "best effort"), so both his PC
client and his SIP phone ring simultaneously. Bob answers on his SIP
phone, and the forked call leg to the PC client is canceled.
F30' 487 (INVITE) Bob's PC Client -> Edge Proxy B
SIP/2.0 487 Request Terminated
Via: SIP/2.0/TCP eb.example.com:5060;branch=z9hG4bKtroubaba
Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Content-Length: 0
F31' 487 (INVITE) Edge Proxy B -> Registrar/Authoritative Proxy B
SIP/2.0 487 Request Terminated
Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Content-Length: 0
6.4. Alice Calls Bob's SIP AOR Using TLS
Bob's registration has already occurred as per Section 6.1.
The third example is identical to the second one, except that Alice
uses TLS as the transport for her connection to her proxy. Such an
arrangement would be common if Alice's UA supported TLS and wanted to
use a single connection to the proxy (as would be the case when using
[RFC5626]). In the example below, Proxy A is also using TLS as a
transport to communicate with Outbound Proxy B, but it is not
necessarily the case.
When using a SIP URI in the Request-URI but TLS as a transport for
sending the request, the Via field indicates TLS. The Route header
field (if present) typically would use a SIP URI (but it could also
be a SIPS URI). The Contact header fields and To and From, however
would also normally indicate a SIP URI.
The call flow would be exactly as per the second example
(Section 6.3). The only difference would be that all the Via header
fields would use TLS Via parameters. The URIs would remain SIP URIs
and not SIPS URIs.
7. Further Considerations
SIP [RFC3261] itself introduces some complications with using SIPS,
for example, when Record-Route is not used. When a SIPS URI is used
in a Contact header field in a dialog-initiating request and Record-
Route is not used, that SIPS URI might not be usable by the other
end. If the other end does not support SIPS and/or TLS, it will not
be able to use it. The last-hop exception is an example of when this
can occur. In this case, using Record-Route so that the requests are
sent through proxies can help in making it work. Another example is
that even in a case where the Contact header field is a SIPS URI, no
Record-Route is used, and the far end supports SIPS and TLS, it might
still not be possible for the far end to establish a TLS connection
with the SIP originating end if the certificate cannot be validated
by the far end. This could typically be the case if the originating
end was using server-side authentication as described below, or if
the originating end is not using a certificate that can be validated.
TLS itself has a significant impact on how SIPS can be used. Server-
side authentication (where the server side provides its certificate
but the client side does not) is typically used between a SIP end-
user device acting as the TLS client side (e.g., a phone or a
personal computer) and its SIP server (proxy or registrar) acting as
the TLS server side. TLS mutual authentication (where both the
client side and the server side provide their respective
certificates) is typically used between SIP servers (proxies,
registrars), or statically configured devices such as PSTN gateways
or media servers. In the mutual authentication model, for two
entities to be able to establish a TLS connection, it is required
that both sides be able to validate each other's certificates, either
by static configuration or by being able to recurse to a valid root
certificate. With server-side authentication, only the client side
is capable of validating the server side's certificate, as the client
side does not provide a certificate. The consequences of all this
are that whenever a SIPS URI is used to establish a TLS connection,
it is expected to be possible for the entity establishing the
connection (the client) to validate the certificate from the server
side. For server-side authentication, [RFC5626] is the recommended
approach. For mutual authentication, one needs to ensure that the
architecture of the network is such that connections are made between
entities that have access to each other's certificates. Record-Route
[RFC3261] and Path [RFC3327] are very useful in ensuring that
previously established TLS connections can be reused. Other
mechanisms might also be used in certain circumstances: for example,
using root certificates that are widely recognized allows for more
easily created TLS connections.
8. Security Considerations
Most of this document can be considered to be security considerations
since it applies to the usage of the SIPS URI.
The "last-hop exception" of [RFC3261] introduced significant
potential vulnerabilities in SIP, and it has therefore been
deprecated by this specification.
Section 26.4.4 of [RFC3261] describes the security considerations for
the SIPS URI scheme. These security considerations also applies
here, as modified by Appendix A.
9. IANA Considerations
This specification registers two new warning codes, namely, 380 "SIPS
Not Allowed" and 381 "SIPS Required". The warning codes are defined
as follows, and have been included in the Warning Codes (warn-codes)
sub-registry of the SIP Parameters registry available from
http://www.iana.org.
380 SIPS Not Allowed: The UAS or proxy cannot process the request
because the SIPS scheme is not allowed (e.g., because there are
currently no registered SIPS contacts).
381 SIPS Required: The UAS or proxy cannot process the request
because the SIPS scheme is required.
Reference: RFC 5630
The note in the Warning Codes sub-registry is as follows:
Warning codes provide information supplemental to the status code
in SIP response messages.
10. Acknowledgments
The author would like to thank Jon Peterson, Cullen Jennings,
Jonathan Rosenberg, John Elwell, Paul Kyzivat, Eric Rescorla, Robert
Sparks, Rifaat Shekh-Yusef, Peter Reissner, Tina Tsou, Keith Drage,
Brian Stucker, Patrick Ma, Lavis Zhou, Joel Halpern, Hisham
Karthabil, Dean Willis, Eric Tremblay, Hans Persson, and Ben Campbell
for their careful review and input. Many thanks to Rohan Mahy for
helping me with the subtleties of [RFC5626].
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.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
[RFC5626] Jennings, C., "Managing Client-Initiated Connections in
the Session Initiation Protocol (SIP)", RFC 5626, October
2009.
11.2. Informative References
[RFC2543] Handley, M., Schulzrinne, H., Schooler, E., and J.
Rosenberg, "SIP: Session Initiation Protocol", RFC 2543,
March 1999.
[RFC3327] Willis, D. and B. Hoeneisen, "Session Initiation Protocol
(SIP) Extension Header Field for Registering Non-Adjacent
Contacts", RFC 3327, December 2002.
[RFC3515] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
[RFC3608] Willis, D. and B. Hoeneisen, "Session Initiation Protocol
(SIP) Extension Header Field for Service Route Discovery
During Registration", RFC 3608, October 2003.
[RFC3725] 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.
[RFC3891] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
Protocol (SIP) "Replaces" Header", RFC 3891,
September 2004.
[RFC3893] Peterson, J., "Session Initiation Protocol (SIP)
Authenticated Identity Body (AIB) Format", RFC 3893,
September 2004.
[RFC3911] Mahy, R. and D. Petrie, "The Session Initiation Protocol
(SIP) "Join" Header", RFC 3911, October 2004.
[RFC4168] Rosenberg, J., Schulzrinne, H., and G. Camarillo, "The
Stream Control Transmission Protocol (SCTP) as a Transport
for the Session Initiation Protocol (SIP)", RFC 4168,
October 2005.
[RFC4244] Barnes, M., "An Extension to the Session Initiation
Protocol (SIP) for Request History Information", RFC 4244,
November 2005.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC5627] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent URIs (GRUU) in the Session Initiation Protocol
(SIP)", RFC 5627, October 2009.
Appendix A. Bug Fixes for RFC 3261
In order to support the material in this document, this section makes
corrections to RFC 3261.
The last sentence of the fifth paragraph of Section 8.1.3.5 is
replaced by:
The client SHOULD retry the request, this time, using a SIP URI
unless the original Request-URI used a SIPS scheme, in which case
the client MUST NOT retry the request automatically.
The fifth paragraph of Section 10.2.1 is replaced by:
If the Address of Record in the To header field of a REGISTER
request is a SIPS URI, then the UAC MUST also include only SIPS
URIs in any Contact header field value in the requests.
In Section 16.7 on p. 112 describing Record-Route, the second
paragraph is deleted.
The last paragraph of Section 19.1 is reworded as follows:
A SIPS URI specifies that the resource be contacted securely.
This means, in particular, that TLS is to be used on each hop
between the UAC and the resource identified by the target SIPS
URI. Any resources described by a SIP URI (...)
In the third paragraph of Section 20.43, the words "the session
description" in the first sentence are replaced with "SIP". Later in
the paragraph, "390" is replaced with "380", and "miscellaneous
warnings" is replaced with "miscellaneous SIP-related warnings".
The second paragraph of Section 26.2.2 is reworded as follows:
(...) When used as the Request-URI of a request, the SIPS scheme
signifies that each hop over which the request is forwarded, until
the request reaches the resource identified by the Request-URI, is
secured with TLS. When used by the originator of a request (as
would be the case if they employed a SIPS URI as the address-of-
record of the target), SIPS dictates that the entire request path
to the target domain be so secured.
The first paragraph of Section 26.4.4 is replaced by the following:
Actually using TLS on every segment of a request path entails that
the terminating UAS is reachable over TLS (by registering with a
SIPS URI as a contact address). The SIPS scheme implies
transitive trust. Obviously, there is nothing that prevents
proxies from cheating. Thus, SIPS cannot guarantee that TLS usage
will be truly respected end-to-end on each segment of a request
path. Note that since many UAs will not accept incoming TLS
connections, even those UAs that do support TLS will be required
to maintain persistent TLS connections as described in the TLS
limitations section above in order to receive requests over TLS as
a UAS.
The first sentence of the third paragraph of Section 26.4.4 is
replaced by the following:
Ensuring that TLS will be used for all of the request segments up
to the target UAS is somewhat complex.
The fourth paragraph of Section 26.4.4 is deleted.
The last sentence of the fifth paragraph of Section 26.4.4 is
reworded as follows:
S/MIME or, preferably, [RFC4474] may also be used by the
originating UAC to help ensure that the original form of the To
header field is carried end-to-end.
In the third paragraph of Section 27.2, the phrase "when the failure
of the transaction results from a Session Description Protocol (SDP)
(RFC 2327 [1]) problem" is deleted.
In the fifth paragraph of Section 27.2, "390" is replaced with "380",
and "miscellaneous warnings" is replaced with "miscellaneous SIP-
related warnings".
Author's Address
Francois Audet
Skype Labs
EMail: francois.audet@skypelabs.com
