SIP F. Audet
Internet-Draft Nortel Networks
Intended status: Informational February 7, 2007
Expires: August 11, 2007
Guidelines for the use of the SIPS URI Scheme in the Session Initiation
Protocol (SIP)
draft-ietf-sip-sips-01
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Copyright Notice
Copyright (C) The IETF Trust (2007).
Abstract
This document providing clarifications and guidelines concerning the
use of SIPS URI scheme in the Session Initiation Protocol (SIP). It
does not make normative changes to SIP. This document also provides
discussion of possible future steps in specification, and the pros
and cons of making such changes.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Meaning of SIPS . . . . . . . . . . . . . . . . . . . . . . . 3
3.1. Scope of SIPS . . . . . . . . . . . . . . . . . . . . . . 5
3.2. Using TLS with SIP instead of SIPS . . . . . . . . . . . . 5
4. Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
4.1. Detection of hop-by-hop security on each hop . . . . . . . 8
4.2. Loose and strict routing . . . . . . . . . . . . . . . . . 9
5. Registration . . . . . . . . . . . . . . . . . . . . . . . . . 11
6. SIPS in a Dialog . . . . . . . . . . . . . . . . . . . . . . . 12
7. Usage of tls transport parameter and TLS Via parameter . . . . 13
8. GRUU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
9. Call Flows . . . . . . . . . . . . . . . . . . . . . . . . . . 15
9.1. Alice Calls Bob's SIPS AOR . . . . . . . . . . . . . . . . 15
9.2. Alice Calls Bob's SIP AOR using TCP . . . . . . . . . . . 24
9.3. Alice Calls Bob's SIP AOR using TLS . . . . . . . . . . . 35
10. Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . 42
11. Security Considerations . . . . . . . . . . . . . . . . . . . 43
12. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 43
13. IAB Considerations . . . . . . . . . . . . . . . . . . . . . . 43
14. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 43
15. References . . . . . . . . . . . . . . . . . . . . . . . . . . 44
15.1. Normative References . . . . . . . . . . . . . . . . . . . 44
15.2. Informational References . . . . . . . . . . . . . . . . . 44
Appendix A. Future steps in specification . . . . . . . . . . . . 45
A.1. Deprecation of Last Hop Exception . . . . . . . . . . . . 45
A.2. True-end-to-end SIP security . . . . . . . . . . . . . . . 46
Appendix B. To-Be-Done . . . . . . . . . . . . . . . . . . . . . 46
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 46
Intellectual Property and Copyright Statements . . . . . . . . . . 48
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1. Introduction
The meaning and usage of the SIPS URI scheme and of TLS [RFC4346] is
underspecified in SIP [RFC3261] and has been the source of confusion
for implementors.
This document provides clarifications and guidelines concerning the
use of the SIPS URI scheme in the Session Initiation Protocol (SIP).
It does not make normative changes to SIP. This document also
provides discussion of possible future steps in specification, and
the pros and cons of making such changes.
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 [RFC2119].
3. Meaning of SIPS
[RFC3261]/19.1 describes a SIPS URI as follows:
A SIPS URI specifies that the resource be contacted securely.
This means, in particular, that TLS is to be used between the UAC
and the domain that owns the URI. From there, secure
communications are used to reach the user, where the specific
security mechanism depends on the policy of the domain.
Section 26.2.2 re-iterates it, with regards to Request-URIs:
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 SIP entity responsible for the domain
portion of the Request-URI, must be secured with TLS; once it
reaches the domain in question it is handled in accordance with
local security and routing policy, quite possibly using TLS for
any last hop to a UAS. 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.
Let's take the classic SIP trapezoid to explain the meaning of a
sips:b@B URI. Instead of using real domain names like example.com
and example.net, logical names like "A" and "B" are used, for
clarity.
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.......................... ...........................
. . . .
. +-------+ . . +-------+ .
. | | . . | | .
. | Proxy |-----TLS---- | Proxy | .
. | A | . . | B | .
. | | . . | | .
. / +-------+ . . +-------+ \ .
. / . . \ .
. / . . \ .
. TLS . . Policy-based .
. / . . \ .
. / . . \ .
. / . . \ .
. +-------+ . . +-------+ .
. | | . . | | .
. | UA a | . . | UA b | .
. | | . . | | .
. +-------+ . . +-------+ .
. Domain A . . Domain B .
.......................... ...........................
SIP trapezoid
In this case, if a@A is sending a request to sips:b@B, the following
will apply:
o TLS must be used between UA a@A and Proxy A
o TLS must be used between Proxy A and Proxy B
o TLS may be used between Proxy B and UA b@B, depending on local
policy.
One may then wonder why TLS is mandatory between UA a@A and Proxy A
but not between Proxy B and UA b@B. The main reason is that [RFC3261]
was written before [I-D.ietf-sip-outbound]. At that time, it was
recognized that in many practical deployments, Proxy B may not be
able to establish a TLS connection with UA b because only Proxy B
would have a certificate to provide and UA b would have none. Since
UA b would be the TLS Server, it would then not be able to accept the
incoming TLS connection. The consequence is that an [RFC3261]-
compliant UAS b, while it may not need to support TLS for incoming
requests, will nevertheless have to support TLS for outgoing requests
as it takes the UAC role. Contrary to what many believe erroneously,
the last-hop exception was not created to allow for using a SIPS URI
to address a UAS that does not support TLS : the last-hop exception
was an attempt to allow for incoming requests TLS when a SIPS URI is
used, and it does not apply to outgoing requests. The rationale for
this was somewhat flawed, and since then, [I-D.ietf-sip-outbound] has
provided a more satisfactory solution to this problem.
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Furthemore, consider the problem of using SIPS inside a dialog. If
a@A sends a request to b@B using a SIPS Request-URI, according to
[RFC3261]/8.1.1.8, then the contact MUST contain a SIPS URI as well.
This means that b@B, upon sending a new Request within the dialog
(e.g., a BYE or re-INVITE), will have to use a SIPS URI. If there is
no Record-Route entry, or if the last Record-Route entry consist of a
SIPS URI, this implies that b@B must understand SIPS in the first
place, and must also support TLS. If the last Record-Route entry
however is a sip URI, then b would be able to send requests without
using TLS (but b would still have to be able to handle SIPS schemes
when parsing the message). In either case, the Request-URI would be
a SIPS URI.
It is therefore RECOMMENDED that implementations do not use the last
hop exception rule and use TLS all the way up to the remote target.
The SIPS scheme implies transitive trust. Obviously, there is
nothing that prevents a proxy to cheat (see [RFC3261]/26.4.4). While
SIPS is useful to request that a resource be contacted securely, it
is not useful as an indication that a resource was in fact contacted
security. Therefore, it is not appropriate to infer that because an
incoming request had a Request-URI (or To header) containing a SIPS
URI, that it necessarily garantees that the request was in fact
transmitted securely on each hop. Some have been tempted to believe
that the SIPS scheme was equivalent to an HTTPS scheme in the sense
that one could provide a visual indication to a user (e.g., a padlock
icon) to the effect that the session is secured. This is obviously
not the case, and one must therefore be careful not to oversell the
meaning of a SIPS URI. There is currently no mechanism to provide an
indication of end-to-end security for SIP. Other mechanisms may
provide a more concrete indication of some level of security. For
example, SIP Identity [RFC4474] describes an authenticated identity
mechanism and a domain-to-domain integrity protection mechanism.
3.1. Scope of SIPS
SIPS means that the SIP resource designated by the target SIPS URI be
contacted securily, using TLS, on each SIP hop (last-hop exception
nothwistanding). It is outside of the scope of this document to
specify what happens when a SIPS URI identifies a UAS resource that
"maps" outside of the SIP network, for example, to other networks
such as the PSTN.
3.2. Using TLS with SIP instead of SIPS
Because SIPS URI imply that requests sent to it be sent over each SIP
hop over TLS (last-hop exception nothwistranding), SIPS URIs are not
suitable for "best-effort TLS": they are only suitable for "TLS-only"
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requests. This is recognized in section [RFC3261]/26.2.2
Users that distribute a SIPS URI as an address-of-record may elect
to operate devices that refuse requests over insecure transports.
If one wants to use "best-effort TLS" for SIP, one just needs to use
a SIP URI, and to send the request over TLS.
Using SIP over TLS is very simple. A UA opens a TLS connection and
uses SIP URIs instead of SIPS URIs for all the headers in a SIP
message (From, To, Request-URI, Contact, Route, etc.). However, when
TLS is used, the Via header MUST indicate TLS.
Similarly, proxies may forward requests using TLS if they can open a
TLS connection, even if the route set used SIP URIs instead of SIPS
URIs. The proxies may insert Record-Route using SIP URIs even if it
uses TLS transport.
Some proxies may have local policies that enforces TLS on all
connections, independently of if SIPS is used or not.
4. Routing
This specification clarifies that SIP and SIPS URIs that are
identical except for the scheme itself (e.g., sip:alice@example.com
and sips:alice@example.com) MUST refer to the same resource. This
requirement is implicit in [RFC3261]/19.1 which states that "Any
resource described by a SIP URI can be "upgraded" to a SIPS URI by
just changing the scheme, if it is desired to communicate with that
resource securily". This does not mean that the SIPS URI will
necessarily be reachable, in particular, if the proxy can not
establish a secure connection to a client or another proxy. This
does not suggest that proxies should arbitrarily "upgrade" SIP URIs
to SIPS URIs. Rather, it means that when a proxy has a legitimate
reason to do so, it MAY upgrade a SIP URI to a SIPS URI.
An example of such a case is when the Contact binding to an AOR is a
SIPS URI and a request was addressed to a SIP Request-URI, the proxy
will "upgrade" by forwarding the request to the endpoint that has
registered with a SIPS contact, as illustrated by message F13 in
Section 9.2 and in Section 9.3. Note that the proxy MUST forward the
request to the UA using a SIP Request-URI and not a SIPS Request-URI
(and it does so by substituting the sip scheme to the sips sheme in
the registered Contact binding). If the proxy did not do this, and
instead used a SIPS Request-URI, then the response (e.g., a 200 to an
INVITE) would have to include a SIPS Contact. That SIPS Contact
would then force the other UA to use a SIPS Contact in any mid-dialog
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request, including the ACK (which wouldn't be possible if that UA did
not support SIPS).
Although not mandated specifically in [RFC3261], the implication is
that a resource described by a SIPS URI can not be "downgraded" to a
SIP URI by just changing the scheme, unless it is the "last hop
exception" described in Section 3. This specification clarifies that
a resource described by a SIPS URI MUST NOT be "downgraded" to a SIP
URI by changing the scheme, or by sending the associated request over
a non secure link, except for cases where the "last-hop exception"
rule is in effect (in which case the Request-URI would be replaced by
a SIP URI).
For example, the sip:bob@example.com and sips:bob@example.com AORs
MUST refer to the same user "Bob" in domain "example.com": the first
URI is the SIP version, and the second one is the SIPS version. From
the point of view of routing, requests to either sip:bob@example.com
and sips:bob@example.com are treated the same way. Location services
are therefore free to map from SIP to SIPS URIs as appropriate (see
[RFC3261]/26.4.4). When Bob registers, it therefore does not really
matter if he is using a SIP or a SIPS AOR, since they both refer to
the same user. At first glance, section [RFC3261]/19.1.4 seems to
contradict this idea by stating that a SIP and a SIPS URI are never
equivalent. Specifically, it says that they are never equivalent for
the purpose of comparing bindings in Contact URIs in REGISTER
requests. The key point is that this statement applies to the
Contact bindings in a registration: it is the association of the
Contact with the AoR that will determine if the user is reachable or
not with a SIPS URI.
Consider this example. If Bob registers with a SIPS contact (e.g.,
sips:bob@bobphone.example.com), the registar and the location service
then knows that Bob (bob@example.com) is reachable at
sips:bob@bobphone.example.com, and at sip:bob@bobphone.example.com.
If a request is sent to AOR sips:bob@example.com, Bob's proxy will
route it to Bob at Request-URI sips:bob@bobphone.example.com. If a
request is sent to AOR sip:bob@examplecom, Bob's proxy will route it
to Bob at Request-URI sip:bob@bobphone.example.com (because of the
"upgrade" scenario described above, and the substitution of SIP to
SIPS).
However, if Bob had registered instead with a SIP Contact instead of
a SIPS contact (e.g., sip:bob@bobphone.example.com), then a request
to AOR sips:bob@example.com would not be routed to Bob, since there
is no SIPS contact for Bob, and "downgrades" from SIPS to SIP are not
allowed.
See Section 9 for illustrative call flows.
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Since upgrading from SIP to SIPS is allowed it other circumstances
(e.g., a user "guessing" a SIPS AOR from a SIP AOR on a business
card), it is quite possible that a request will be rejected with
response code 416 (either because TLS or SIPS is not supported).
When 416 is received, the request MAY be re-attempted with a SIP URI,
but the user SHOULD be informed.
Although "downgrading" from SIPS to SIP is disallowed, it is possible
that a redirect server or UAS sends a 3XX response to a request to a
SIPS URI with a Contact containing a SIP URI. [RFC3261]/8.1.3.4
recommends that if the UAC decide to recurse to the SIP URI, it
SHOULD inform the user. When a proxy is handling the 3XX, it can
obviously not indicate anything to the user that it is being
redirected from SIPS to SIP: therefore, proxies MUST NOT recurse on
the Contact. The proxy may do one of the following:
o Forward the 3XX to the UAC instead of recursing, in order to allow
for the UAC to take the appropriate action;
o Reject the call with a 403 "Forbidden".
4.1. Detection of hop-by-hop security on each hop
The presence of a SIPS Request-URI does not necessarily indicate that
the request was sent securely on each hop. As described in
[RFC3261]/26.4.4, a proxy may legitimately retarget a request from
SIP to SIPS. Therefore, a UAS MUST NOT assume on the basis of the
Request-URI alone that SIPS was used for the entire request path.
So how does a UAS know if the SIPS was used for the entire request
path to secure the request end-to-end? Effectively, the UAS can not
know for sure. However, [RFC3261]/26.4.4 recommends how a UAS may
make some checks to validate the security. Here is a summary of a
potential algorithm:
o If the URI in the To header is a SIPS URI and the Request-URI is a
SIPS, then the dialog is "tentatively" secure. See below.
o If the URI in the To header is SIPS and the Request-URI is SIP and
there is some other security mechanism (e.g., IPsec) securing the
last hop, then the dialog may be "tentatively" secure. See below.
o Otherwise the dialog is insecure.
o If the dialog was "tentatively" secure, it is RECOMMENDED that the
security be checked by checking both the Via headers and the
Record-route, as described in [RFC3261]/26.4.4.
o Additionaly, the History-Info header [RFC4244] may be inspected
for detecting retargeting between SIP and SIPS.
Again, it should be restated that all the checking may be
circumvented by any proxy on the path that does not follow the rules
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and recommendations of this document and of [RFC3261].
Proxies MAY have their own policy regarding routing of requests to
SIP or SIPS URIs. For example, a proxy in some environment may be
configured to only route SIPS. Some proxies MAY be configured to
detect uncompliancies and reject unsecure requests. For example, it
could inspect Request-URIs, Path, Record-Route, To, From, Contacts
and Via headers to enforce SIPS.
[RFC3261]/26.4.4 also explains that S/MIME may also be used by the
originating UAC to ensure that the original form of the To header
field is carried end-to-end. While not specifically mentioned in
[RFC3261]/26.4.4, this is meant to imply that [RFC3893] would be used
to "tunnel" important headers (such as To and From) in an encrypted
and signed S/MIME body, replicating the information in the SIP
message, and allowing the UAS to validate the content of those
important headers. While this approach is certainly legal, another
approach is to use the SIP Identity mechanism defined in [RFC4474].
SIP Identity creates a signed identity digest which includes, amongst
other things, the AOR of the sender (from the From header) and the
AOR of the original destination (from the To header). It is
RECOMMENDED that a UAC use the mechanism in [RFC4474] instead of the
one defined in [RFC3893].
4.2. Loose and strict routing
Using strict or loose routing has a huge impact on sips and TLS.
Some of the advantages of using loose routing have been discussed in
Section 3, regarding mid-dialog requests.
When a proxy inserts a Record-Route entry, it must take care in using
the proper scheme so that further in-dialog requests are sent to the
proper URI. This is particularly important when a proxy changes the
Request-URI of a request from SIPS to SIP (when the last hop
exception rule is used) or from SIP to SIPS. [RFC3261] sections 16.6
and 16.7 describe how this can be done by having the proxy modifying
the Record-Route in the response. However, as described in
[RFC3608], this is problematic. This document therefore uses the
procedure of [RFC3608], and instead of following the procedure in
[RFC3261], proxies that are inserting Record-Route or Path header
field URIs SHOULD record not one but two route URIs when processing
the request in the case where the scheme is changed. The first value
recorded indicates the schme of the receiving interface, and the
second indicates the scheme of the sending interface. When
processing the response, no modification of the recorded route is
required. This optimization provides for fully invertible routes
that can be effectively used in construction of service routes.
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If the Request-URI or the topmost Route header on the receiving
interface is SIPS and the Request-URI on the sending interface is
SIP, then the first value recorded uses a SIPS URI and the second
value indicates a SIP URI. It is illustrated as follows:
UA a Proxy UA b
-------REQUEST sips:-------->-------REQUEST sip:--------->
Record-Route: <sip:p;lr>,
<sips:p;lr>
<------Response sips:-------<-------Response sip:---------
Record-Route: <sip:p;lr> Record-Route: <sip:p;lr>,
<sips:p;lr> <sips:p;lr>
Record routing from SIPS to SIP
If the Request-UIR on the receiving interface is SIP and the Request-
URI on the sending interface is SIPS, then the first value recorded
uses a SIP URI and the second value indicates a SIPS URI. It is
illustrated as follows:
UA a Proxy UA b
-------REQUEST sip:--------->-------REQUEST sips:-------->
Record-Route: <sips:p;lr>,
<sip:p;lr>
<------Response sip:--------<-------Response sips:--------
Record-Route: <sips:p;lr> Record-Route: <sips:p;lr>,
<sip:p;lr> <sip:p;lr>
Record routing from SIP to SIPS
Note that the same rules apply to the Path Header [RFC3327].
When a UAC is using a Service-Route (e.g., as in [RFC3608]), and
sending a request to a SIPS Request-URI, it MUST ensure that the
Route header URIs it includes are all SIPS URIs. If the service
route included SIP URI, the UAC MUST upgrade the SIP URIs to SIPS
URIs simply by changing the scheme from "sip" to "sips" before
sending the request. Note that this allows for configuring or
discovering one Service Route with all SIP URIs and allowing sending
requests to both SIP and SIPS URIs.
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5. Registration
This section describes the registration procedures of SIPS versus SIP
Contacts that follows from the discussion in Section 4 and clarifies
[RFC3261].
The USC registers either a SIPS or a SIP AOR. From a routing
perspective, it does not matter which one is used for registration as
they identify the same resource.
If all the Contacts are SIPS, a SIPS AOR MUST also be used by the
UAC. If at least one of the Contacts is SIP or is neither SIP nor
SIPS (e.g., mailto, tel, http, https), a SIP AOR MUST also be used by
the UAC. However, the UAS (the Registrar), must treat the SIP and
SIPS schemes of the AOR the same way (i.e., it must not care if it is
SIP or SIPS). Those are mechanical rules with no influence on
routing.
Furthermore, it is a matter of local policy for a UA to accept
incoming requests addressed to a URI scheme that does not correspond
to what it used for registration. For example, a UA with a policy of
"always SIPS" MUST address the Registrar using a SIPS Request-URI
over TLS, MUST register with a SIPS Contact, and MUST reject requests
addressed to a SIP Request-URI with 403 "Forbidden". A UA with a
policy of "best-effort SIPS" MUST address the Registrar using a SIPS
Request-URI over TLS, MUST register with a SIPS Contact, and MUST
accept requests addressed to either SIP or SIPS Request-URIs. A UA
with a policy of "No SIPS" MUST address the Registrar using a SIP
Request-URI, MAY or MAY NOTt use TLS, MUST register with a SIP AOR
and a SIP Contact, and MUST accept requests addressed to a SIP
Request-URI.
If proxies are present in the path between the UA and the registrar,
they may insert the Path header [RFC3327] if the UAC has indicated
support for it with a supported header.
A registrar MUST only accept a binding to a SIPS Contact if all the
appropriate URIs are of the SIPS scheme: i.e., the Request-URI, the
AOR (i.e., To header), the From header, the Contacts and all the Path
headers. If the URIs are not of the proper SIPS scheme, it must
reject the REGISTER with a 403 "Forbidden".
The usage of the "transport" URI parameter in Contacts in
registration is of dubious usefulness. The assumption is that a UAC
may choose one transport for the registration itself, and a different
transport for receiving requests. Using the transport URI parameters
also results in some complex problems. For example, should all the
transports be listed as separate contacts (e.g, udp, tcp, sctp, tls
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over tcp, tls over sctp)? If so, there is no way to signal tls over
sctp defined yet. Furthermore, how should they be prioritized using
a q-value? If so, it is possible that certain proxies will interpret
this as a forking scenario and they might decide to send one incoming
request per transport. Another issue is what happens if a UAC
fetches bindings by sending an empty REGISTER message. Would the
proxy respond with one or all the possible transport? All this would
generate unwarranted complexity.
It is therefore RECOMMENDED that UACs do not use any transport URI
parameters in Contacts in REGISTER.
For backward compatibility, a registrar should accept a REGISTER
message with a transport URI parameter in the Contact. It is
recommended that a registrar ignores that parameter, i.e., that it
will not influence routing.
However, a registrar MUST record the scheme of the Contact.
6. SIPS in a Dialog
There MUST be only one Contact in any request resulting in the
establishment of a dialog (e.g., INVITE, SUBSCRIBE, REFER).
As mandated by [RFC3261]/8.1.1.8, in a request, if the Request-URI or
top Route header field value contains a SIPS URI, the Contact header
field MUST contain a SIPS URI as well. Furthermore, as mandated by
[RFC3261]/12.1.1, if the request that initiated the dialog contained
a SIPS URI in the Request-URI or in the top Record-Route header field
value, if there was any, or the Contact header field if there was no
Record-Route header field, the Contact header filed in the response
MUST be a SIPS URI.
If a UAS (or proxy acting on behalf of a UAS) does not support SIPS,
it MUST reject a request to a SIPS Request-URI with response code 416
"Unsupported URI scheme". Upon receiving a 416 a UAC SHOULD NOT re-
attempt the request by automatically replacing the SIPS scheme with a
SIP scheme. If the UAC does re-attempt the call with a SIP URI, it
SHOULD inform to the user that the security level is downgraded.
If a UAS (or proxy acting on behalf of a UAS) does not want to be
contacted with a SIP UI, it MAY reject a request to a SIP Request-URI
with response code 403 "Forbidden", or it MAY redirect the request to
a SIPS URI with 301 "Moved permanently" or 302 "Moved temporarily" as
appropriate.
If the Request-URI in a request is a SIP URI, then the UAC needs to
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be careful about what to use in the Contact (in case Record-Route is
not used for this hop). If the Contact was a SIPS URI, it would mean
that it would only accept mid-dialog requests that are over secure
transport end-to-end. Since the Request-URI is in this case a SIP
URI, it is quite possible that the UA sending a request to that URI
may not be able to send requests to SIPS URIs. It is therefore
RECOMMENDED that in this case, the Contact be a SIP URI, even if the
request is sent over a secure transport (e.g., the first hop could be
re-using a TLS connection to the proxy as would be the case with
[I-D.ietf-sip-outbound]).
When a target refresh occurs within a dialog (e.g., re-INVITE,
UPDATE), unless there is a need to change it, the UAC SHOULD include
a Contact header with a SIPS URI if the original request used a SIPS
Request-URI.
7. Usage of tls transport parameter and TLS Via parameter
[RFC3261]/26.2.2 makes it clear that the use of the "transport=tls"
URI transport parameter in SIPS or SIP URIs has been deprecated:
Note that in the SIPS URI scheme, transport is independent of TLS,
and thus "sips:alice@atlanta.com;transport=tcp" and
"sips:alice@atlanta.com;transport=sctp" are both valid (although
note that UDP is not a valid transport for SIPS). The use of
"transport=tls" has consequently been deprecated, partly because
it was specific to a single hop of the request. This is a change
since RFC 2543.
However, the "tls" parameter has not been eliminated from the ABNF in
[RFC3261]/25, and [RFC3261]/26.2.1 has a vague reference to it. This
has been a source of confusion. Those omissions are errors in
[RFC3261].
NOTE: This needs to be in corrected in [RFC3261] in another document
making normative changes to [RFC3261].
This specification clarifies that the "transport=tls" parameter MUST
NOT be used.
However, for backward compatibility, if a "transport=tls" parameter
is received, it should be interpreted as per the following
guidelines:
o [RFC3261]/16.7 states the transport parameter (e.g., with tcp or
udp) SHOULD NOT be used in Record-Route unless it has knowledge
that the next upstream element that will be in the path of
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subsequent supports this transport. Generally, it is recommended
that the transport parameter never be used in a Record-Route,
Route or Path header. Since the transport=tls URI parameter has
been deprecated, it MUST NOT be used in Route, Record-Route or
Path headers, and MUST be ignored.
o In a Contact in a dialog, it MAY be interpreted as a request to
send incoming mid-dialog requests using TLS. Note that this would
only have a significance if [I-D.ietf-sip-outbound] and Record-
Route are not used, and if that URI is nevertheless reachable with
TLS which is extremely unlikely. If it was the case that it was
reachable with TLS, say because there is an active TLS connection,
then that connection could be re-used anyways, regardless of the
presense of the transport parameter. It is RECOMMENDED that the
"transport=tls" parameter be ignored by the UAS.
o In a Contact in a REGISTER, it tells the registrar that the UAC is
reachable through TLS. If the registrar and proxy are co-located,
and are the proxy of that UAC, it tells what is already known
because the request was sent over TLS (i.e., that it is reachable
using TLS), and is therefore redundant. If the registrar is not
co-located with the proxy, then it is useless because
transport=tls is hop-by-hop and therefore not applicable in this
case. The transport=tls parameter MUST therefore be ignored.
o In a Request-URI, the transport parameter is problematic. On the
last hop, it is useless because the transport is evident. Before
being resolved to the last hop (with loose routing), it is not
clear what it would mean (hop-by-hop?). A proxy MUST ignore the
"transport=tls" parameter in a Request-URI.
o In a Contact in a 3XX response, it would essentially mean a
request to attempt to re-send the request, using TLS transport.
Since the transport=tls parameter only has local significance, it
will only be successful if the 3XX is recursed by the last hop.
It MAY be ignored by the recursing entity, or the recursing entity
MAY re-attempt the request using TLS transport.
For Via headers, the following transport protocol are defined: "UDP",
"TCP", "TLS", "SCTP", and "TLS-SCTP" [RFC4168].
8. GRUU
When a GRUU [I-D.ietf-sip-gruu] is assigned to an instance ID/AOR
pair, both SIP and SIPS GRUUs will be assigned. When a GRUU is
obtained through registration, if the To header in the REGISTER
request contains a SIP URI, the SIP version of the GRUU is returned.
If the To header filed in the REGISTER request contains a SIPS URI,
the SIPS version of the GRUU is returned.
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9. Call Flows
In the following examples, Bob has two clients, one is a SIP PC
client running on his computer, and the other one is a SIP Phone.
The PC client does not support SIPS (and does not support TLS either)
and consequently only registers with a SIP address. The SIP phone
however does support SIPS and TLS, and consequently registers with a
SIPS address. Both of Bob's devices are going through Outbound Proxy
B, and consequently, they include a Route header indicating Proxy B.
Proxy B removes the Route header corresponding to itself, and adds
itself in a Path header.
After registration, there are 2 contact bindings associated with
Bob's AOR of bob@example.com: sips:bob@bobphone.example.com and
sip:bob@bobpc.example.com.
Alice then calls Bob through her own Oubound Proxy A, including a
Route header for Proxy A. Proxy A locates Bob's domain example.com.
In this example, that domain is co-located with Bob's outbound proxy,
but it could easily have been a separate proxy. Outbound Proxy A
removes the Route header corresponding to itself, and inserts itself
in the Record-Route and forwards the request to Proxy B.
The following subsections illustrates three examples. In the first
one, Alice calls Bob using Bob's SIPS URI. In the second one, Alice
calls Bob's SIP AOR using TCP transport. In the third example, Alice
calls Bob's SIP AOR using TLS transport.
9.1. Alice Calls Bob's SIPS AOR
In this first example, Alice calls Bob's SIPS address
(sips:bob@example.com). Proxy B consults the binding in the
registration database, and finds the 2 Contact bindings. Alice had
addressed Bob with a SIPS Request-URI (sips:bob@example.com), so
Proxy B determines that the calls needs to be routed only to a SIPS
Contact, and therefore the request is only sent to
sips:bob@bobphone.example.com. Proxy B inserts itself in the Record-
Route. Bob answers.
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Outbound Outbound
Bob@bobpc Proxy B Registrar Proxy A Alice
| | | | |
| REGISTER F1 | | | |
|---------------->|REGISTER F2 | | |
| |----------->| | |
| | 200 F3 | | |
| 200 F4 |<-----------| | |
|---------------->| | | |
| | | | |
| Bob@phone | | | |
| | | | | |
| |REGISTER F5 | | | |
| |----------->|REGISTER F6 | | |
| | |----------->| | |
| | | 200 F7 | | |
| | 200 F8 |<-----------| | |
| |----------->| | | |
| | | | INVITE F9 |
| | | INVITE F11 |<-----------|
| | INVITE F13 |<------------------------| 100 F10 |
| |<-----------| 100 F12 |----------->|
| | 100 F14 |------------------------>| |
| |----------->| | |
| | 200 F15 | | |
| |----------->| 200 F16 | |
| | |------------------------>| 200 F17 |
| | | |----------->|
| | | | ACK F18 |
| | | ACK F19 |<-----------|
| | ACK F20 |<------------------------| |
| |<-----------| | |
Alice Calls Bob's SIPS AOR
Message details
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F1 REGISTER Bob's PC Client -> Proxy B
REGISTER sip:registrar.example.com SIP/2.0
Via: SIP/2.0/TCP bobspc.example.com:5060;branch=z9hG4bKnashds
Max-Forwards: 70
To: Bob <sip:bob@example.com>
From: Bob <sip:bob@example.com>;tag=456248
Call-ID: 843817637684230@998sdasdh09
CSeq: 1826 REGISTER
Supported: path
Route: <sip:proxyb.example.com;lr>
Contact: <sip:bob@bobpc.example.com>
;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
;reg-id=1
Expires: 7200
Content-Length: 0
F2 REGISTER Proxy B -> Registrar
REGISTER sip:registrar.example.com SIP/2.0
Via: SIP/2.0/TCP proxyb.example.com:5060;branch=z9hG4bK87asdks7
Via: SIP/2.0/TCP bobspc.example.com:5060;branch=z9hG4bKnashds
Max-Forwards: 69
To: Bob <sip:bob@example.com>
From: Bob <sip:bob@example.com>;tag=456248
Call-ID: 843817637684230@998sdasdh09
CSeq: 1826 REGISTER
Supported: path
Path: <sip:laksdyjanseg237+fsdf@proxyb.example.com;lr>
Contact: <sip:bob@bobpc.example.com>
;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
;reg-id=1
Expires: 7200
Content-Length: 0
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F3 200 (REGISTER) Registrar -> Proxy B
SIP 2.0 200 OK
Via: SIP/2.0/TCP proxyb.example.com:5060;branch=z9hG4bK87asdks7
Via: SIP/2.0/TCP bobspc.example.com:5060;branch=z9hG4bKnashds
To: Bob <sip:bob@example.com>;tag=2493K59K9
From: Bob <sip:bob@example.com>;tag=456248
Call-ID: 843817637684230@998sdasdh09
CSeq: 1826 REGISTER
Supported: outbound
Path: <sip:laksdyjanseg237+fsdf@proxyb.example.com;lr>
Contact: <sip:bob@bobphone.example.com>
;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
;reg-id=1
;expires=7200
Date: Mon, 12 Jun 2006 16:43:12 GMT
Content-Length: 0
F4 200 (REGISTER) Proxy B -> Bob's PC Client
SIP 2.0 200 OK
Via: SIP/2.0/TCP bobspc.example.com:5060;branch=z9hG4bKnashds
To: Bob <sip:bob@example.com>;tag=2493K59K9
From: Bob <sip:bob@example.com>;tag=456248
Call-ID: 843817637684230@998sdasdh09
CSeq: 1826 REGISTER
Supported: outbound
Path: <sip:laksdyjanseg237+fsdf@proxyb.example.com;lr>
Contact: <sip:bob@bobphone.example.com>
;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
;reg-id=1
;expires=7200
Date: Mon, 12 Jun 2006 16:43:12 GMT
Content-Length: 0
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F5 REGISTER Bob's Phone -> Proxy B
REGISTER sips:registrar.example.com SIP/2.0
Via: SIP/2.0/TLS bobphone.example.com:5061;branch=z9hG4bK9555
Max-Forwards: 70
To: Bob <sips:bob@example.com>
From: Bob <sips:bob@example.com>;tag=90210
Call-ID: faif9a@qwefnwdclk
CSeq: 12 REGISTER
Supported: path
Route: <sips:proxyb.example.com;lr>
Contact: <sips:bob@bobphone.example.com>
;+sip.instance="<urn:uuid:6F85D4E3-E8AA-46AA-B768-BF39D5912143>"
;reg-id=1
Expires: 7200
Content-Length: 0
F6 REGISTER Proxy B -> Registrar
REGISTER sips:registrar.example.com SIP/2.0
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bK876354
Via: SIP/2.0/TLS bobphone.example.com:5061;branch=z9hG4bK9555
Max-Forwards: 69
To: Bob <sips:bob@example.com>
From: Bob <sips:bob@example.com>;tag=90210
Call-ID: faif9a@qwefnwdclk
CSeq: 12 REGISTER
Supported: path
Path: <sips:psodkfsj+34+kkls@proxyb.example.com;lr>
Contact: <sips:bob@bobphone.example.com>
;+sip.instance="<urn:uuid:6F85D4E3-E8AA-46AA-B768-BF39D5912143>"
;reg-id=1
Expires: 7200
Content-Length: 0
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F7 200 (REGISTER) Registrar -> Proxy B
SIP 2.0 200 OK
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bK876354
Via: SIP/2.0/TLS bobphone.example.com:5061;branch=z9hG4bK9555
To: Bob <sips:bob@example.com>;tag=5150
From: Bob <sips:bob@example.com>;tag=90210
Call-ID: faif9a@qwefnwdclk
CSeq: 12 REGISTER
Supported: outbound
Path: <sips:psodkfsj+34+kkls@proxyb.example.com;lr>
Contact: <sips:bob@bobphone.example.com>
;+sip.instance="<urn:uuid:6F85D4E3-E8AA-46AA-B768-BF39D5912143>"
;reg-id=1
;expires=7200
Date: Mon, 12 Jun 2006 16:43:50 GMT
Content-Length: 0
F8 200 (REGISTER) Proxy B -> Bob's Phone
SIP 2.0 200 OK
Via: SIP/2.0/TLS bobphone.example.com:5061;branch=z9hG4bK9555
To: Bob <sips:bob@example.com>;tag=5150
From: Bob <sips:bob@example.com>;tag=90210
Call-ID: faif9a@qwefnwdclk
CSeq: 12 REGISTER
Supported: outbound
Path: <sips:psodkfsj+34+kkls@proxyb.example.com;lr>
Contact: <sips:bob@bobphone.example.com>
;+sip.instance="<urn:uuid:6F85D4E3-E8AA-46AA-B768-BF39D5912143>"
;reg-id=1
;expires=7200
Date: Mon, 12 Jun 2006 16:43:50 GMT
Content-Length: 0
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F9 INVITE Alice -> Proxy A
INVITE sips:bob@example.com SIP/2.0
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 70
To: Bob <sips:bob@example.com>
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Route: <sips:proxya.example.net;lr>
Contact: <sips:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
F10 100 (INVITE) Proxy A -> Alice
SIP 2.0 100 Trying
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 70
To: Bob <sips:bob@example.com>
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Content-Length: 0
F11 INVITE Proxy A -> Proxy B
INVITE sips:bob@example.com SIP/2.0
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 69
To: Bob <sips:bob@example.com>
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sips:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
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F12 100 (INVITE) Proxy B -> Proxy A
SIP 2.0 100 Trying
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sips:bob@example.com>
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Content-Length: 0
F13 INVITE Proxy B -> Bob's Phone
INVITE sips:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 68
To: Bob <sips:bob@example.com>
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sips:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sips:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
F14 100 (INVITE) Bob's Phone -> Proxy B
SIP 2.0 100 Trying
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sips:bob@example.com>
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Content-Length: 0
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F15 200 (INVITE) Bob's Phone -> Proxy B
SIP 2.0 200 OK
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sips:bob@example.com>;tag=5551212
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sips:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sips:bob@bobphone.example.com>
Content-Length: 0
F16 200 (INVITE) Proxy B -> Proxy A
SIP 2.0 200 OK
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sips:bob@example.com>;tag=5551212
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sips:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sips:bob@bobphone.example.com>
Content-Length: 0
F17 200 (INVITE) Proxy A -> Alice
SIP 2.0 200 OK
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sips:bob@example.com>;tag=5551212
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sips:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sips:bob@bobphone.example.com>
Content-Length: 0
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F18 ACK Alice -> Proxy A
ACK sips:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKksdjf
Max-Forwards: 70
To: Bob <sips:bob@example.com>;tag=5551212
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 ACK
Route: <sips:KFndf+47KsFH@proxya.example.net;lr>,
<sips:UJH-hUdvb65@proxyb.example.com;lr>
Content-Lenght: 0
F19 ACK Proxy A -> Proxy B
ACK sips:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKplo7hy
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKksdjf
Max-Forwards: 69
To: Bob <sips:bob@example.com>;tag=5551212
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 ACK
Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>
Content-Lenght: 0
F20 ACK Proxy B -> Bob's Phone
ACK sips:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bK8msdu2
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKplo7hy
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKksdjf
Max-Forwards: 68
To: Bob <sips:bob@example.com>;tag=5551212
From: Alice <sips:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 ACK
Content-Lenght: 0
9.2. Alice Calls Bob's SIP AOR using TCP
In the second example, Alice calls Bob's SIP address instead
(sip:bob@example.com), and she uses TCP as a transport. Proxy B
consults the binding in the registration database, and finds the 2
Contact bindings. Alice had addressed Bob with a SIP Request-URI
(sip:bob@example.com), so Proxy B determines that the calls needs to
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be routed both to the SIP Contact and the SIPS Contact, and therefore
the request is forked sent to sip:bob@boppc.example.com and
sips:bob@bobphone.example.com. Proxy B inserts itself 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.
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Outbound Outbound
Bob@bobpc Proxy B Registrar Proxy A Alice
| | | | |
| REGISTER F1 | | | |
|---------------->|REGISTER F2 | | |
| |----------->| | |
| | 200 F3 | | |
| 200 F4 |<-----------| | |
|---------------->| | | |
| | | | |
| Bob@phone | | | |
| | | | | |
| |REGISTER F5 | | | |
| |----------->|REGISTER F6 | | |
| | |----------->| | |
| | | 200 F7 | | |
| | 200 F8 |<-----------| | |
| |----------->| | | |
| | | INVITE F9 |
| | INVITE F11 |<-----------|
| INVITE F13' |<------------------------| 100 F10 |
|<----------------| 100 F12 |----------->|
| 100 F14' |------------------------>| |
|---------------->| | |
| 180 F15' | | |
|---------------->| 180 F16' | |
| |------------------------>| 180 F17' |
| | INVITE F13 | |----------->|
| |<-----------| | |
| | 100 F14 | | |
| |----------->| | |
| | 200 F15 | | |
| |----------->| 200 F16 | |
| | |------------------------>| 200 F17 |
| | | |----------->|
| | | | ACK F18 |
| | | ACK F19 |<-----------|
| | ACK F20 |<------------------------| |
| |<-----------| | |
| | | |
| CANCEL F20' | | |
|<----------------| | |
| 200 F21' | | |
|---------------->| | |
| 487 F22' | | |
|---------------->| | |
Alice Calls Bob's SIP AOR
Audet Expires August 11, 2007 [Page 26]
Internet-Draft SIPS Guidelines February 2007
Messages F1-F8 are identical to the ones in Section 9.1. The other
messages are as follows.
F9 INVITE Alice -> Proxy A
INVITE sip:bob@example.com SIP/2.0
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
Max-Forwards: 70
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Route: <sip:proxya.example.net;lr>
Contact: <sip:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
F10 100 (INVITE) Proxy A -> Alice
SIP 2.0 100 Trying
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
Max-Forwards: 70
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Content-Length: 0
Audet Expires August 11, 2007 [Page 27]
Internet-Draft SIPS Guidelines February 2007
F11 INVITE Proxy A -> Proxy B
INVITE sip:bob@example.com SIP/2.0
Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
Max-Forwards: 69
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
F12 100 (INVITE) Proxy B -> Proxy A
SIP 2.0 100 Trying
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
Audet Expires August 11, 2007 [Page 28]
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F13' INVITE Proxy B -> Bob's PC Client
INVITE sip:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TCP proxyb.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
Max-Forwards: 68
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
F14' 100 (INVITE) Bob's PC Client -> Proxy B
SIP 2.0 100 Trying
Via: SIP/2.0/TCP proxyb.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
Audet Expires August 11, 2007 [Page 29]
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F15' 180 (INVITE) Bob's PC Client -> Proxy B
SIP 2.0 180 Ringing
Via: SIP/2.0/TCP proxyb.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>;tag=963258
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobpc.example.com>
Content-Length: 0
F16' 180 (INVITE) Proxy B -> Proxy A
SIP 2.0 180 Ringing
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>;tag=963258
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobpc.example.com>
Content-Length: 0
F17' 180 (INVITE) Proxy A -> Alice
SIP 2.0 180 Ringing
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
To: Bob <sip:bob@example.com>;tag=963258
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobpc.example.com>
Content-Length: 0
Audet Expires August 11, 2007 [Page 30]
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F13 INVITE Proxy B -> Bob's Phone
INVITE sip:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba.1
Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
Max-Forwards: 68
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
F14 100 (INVITE) Bob's Phone -> Proxy B
SIP 2.0 100 Trying
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba.1
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
Audet Expires August 11, 2007 [Page 31]
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F15 200 (INVITE) Bob's Phone -> Proxy B
SIP 2.0 200 OK
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba.1
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>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobphone.example.com>
Content-Length: 0
F16 200 (INVITE) Proxy B -> Proxy A
SIP 2.0 200 OK
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>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobphone.example.com>
Content-Length: 0
Audet Expires August 11, 2007 [Page 32]
Internet-Draft SIPS Guidelines February 2007
F17 200 (INVITE) Proxy A -> Alice
SIP 2.0 200 OK
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobphone.example.com>
Content-Length: 0
F18 ACK Alice -> Proxy A
ACK sip:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
Max-Forwards: 70
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 ACK
Route: <sip:KFndf+47KsFH@proxya.example.net;lr>,
<sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sips:UJH-hUdvb65@proxyb.example.com;lr>
Content-Lenght: 0
F19 ACK Proxy A -> Proxy B
ACK sip:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
Max-Forwards: 69
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 ACK
Route: <sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sips:UJH-hUdvb65@proxyb.example.com;lr>
Content-Lenght: 0
Audet Expires August 11, 2007 [Page 33]
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F20 ACK Proxy B -> Bob's Phone
ACK sip:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba.1
Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
Max-Forwards: 68
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 ACK
Content-Lenght: 0
F20' CANCEL Proxy B -> Bob's PC Client
CANCEL sip:bob@bobpc.example.com SIP/2.0
Via: SIP/2.0/TCP proxyb.example.com:5060;branch=z9hG4bKbalouba.2
Max-Forwards: 70
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 CANCEL
Content-Lenght: 0
F21' 200 (CANCEL) Proxy B -> Bob's PC Client
SIP 2.0 200 OK
Via: SIP/2.0/TCP proxyb.example.com:5060;branch=z9hG4bKbalouba.2
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 CANCEL
Content-Lenght: 0
Audet Expires August 11, 2007 [Page 34]
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F22' 487 (INVITE) Proxy B -> Bob's PC Client
SIP 2.0 487 Request Terrminated
Via: SIP/2.0/TCP proxyb.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
9.3. Alice Calls Bob's SIP AOR using TLS
The third example is identical to the second one, except that Alice
uses TLS as the transport for her connection to her outbound proxy.
Such an arrangement would be common if Alice's UA supported TLS and
wanted to use a single connection to the outbound proxy (as would be
the case when using [I-D.ietf-sip-outbound]). In the example below,
Outbound proxy A is also using TLS as a transport to communicate with
Outbound proxy B, but it is not necessarily the case.
It should be noted that 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 (if present) typically would use a SIP URI
(but it could also be a SIPS URI). The Contacts, To and From however
would also normally indicate a SIP URI.
The call flow would be exactly as per the second example.
Messages F1-F8 and F20'-F22' are identical to the ones in
Section 9.2. The other messages are as follows.
Audet Expires August 11, 2007 [Page 35]
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F9 INVITE Alice -> Proxy A
INVITE sip:bob@example.com SIP/2.0
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 70
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Route: <sip:proxya.example.net;lr>
Contact: <sip:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
F10 100 (INVITE) Proxy A -> Alice
SIP 2.0 100 Trying
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 70
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Content-Length: 0
F11 INVITE Proxy A -> Proxy B
INVITE sip:bob@example.com SIP/2.0
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 69
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
Audet Expires August 11, 2007 [Page 36]
Internet-Draft SIPS Guidelines February 2007
F12 100 (INVITE) Proxy B -> Proxy A
SIP 2.0 100 Trying
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;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
F13' INVITE Proxy B -> Bob's PC Client
INVITE sip:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TCP proxyb.example.com:5060;branch=z9hG4bKbalouba.2
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 68
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
F14' 100 (INVITE) Bob's PC Client -> Proxy B
SIP 2.0 100 Trying
Via: SIP/2.0/TCP proxyb.example.com:5060;branch=z9hG4bKbalouba.2
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;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
Audet Expires August 11, 2007 [Page 37]
Internet-Draft SIPS Guidelines February 2007
F15' 180 (INVITE) Bob's PC Client -> Proxy B
SIP 2.0 180 Ringing
Via: SIP/2.0/TCP proxyb.example.com:5060;branch=z9hG4bKbalouba.2
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sip:bob@example.com>;tag=963258
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobpc.example.com>
Content-Length: 0
F16' 180 (INVITE) Proxy B -> Proxy A
SIP 2.0 180 Ringing
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sip:bob@example.com>;tag=963258
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobpc.example.com>
Content-Length: 0
F17' 180 (INVITE) Proxy A -> Alice
SIP 2.0 180 Ringing
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sip:bob@example.com>;tag=963258
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobpc.example.com>
Content-Length: 0
Audet Expires August 11, 2007 [Page 38]
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F13 INVITE Proxy B -> Bob's Phone
INVITE sip:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba.1
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 68
To: Bob <sip:bob@example.com>
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:alice@alice-1.example.net>
Content-Type: application/sdp
Content-Length: {as per SDP}
{SDP not shown}
F14 100 (INVITE) Bob's Phone -> Proxy B
SIP 2.0 100 Trying
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba.1
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;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
Audet Expires August 11, 2007 [Page 39]
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F15 200 (INVITE) Bob's Phone -> Proxy B
SIP 2.0 200 OK
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba.1
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobphone.example.com>
Content-Length: 0
F16 200 (INVITE) Proxy B -> Proxy A
SIP 2.0 200 OK
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobphone.example.com>
Content-Length: 0
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F17 200 (INVITE) Proxy A -> Alice
SIP 2.0 200 OK
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 INVITE
Record-Route: <sips:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sip:KFndf+47KsFH@proxya.example.net;lr>
Contact: <sip:bob@bobphone.example.com>
Content-Length: 0
F18 ACK Alice -> Proxy A
ACK sip:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 70
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 ACK
Route: <sip:KFndf+47KsFH@proxya.example.net;lr>,
<sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sips:UJH-hUdvb65@proxyb.example.com;lr>
Content-Lenght: 0
F19 ACK Proxy A -> Proxy B
ACK sip:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 69
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 ACK
Route: <sip:UJH-hUdvb65@proxyb.example.com;lr>,
<sips:UJH-hUdvb65@proxyb.example.com;lr>
Content-Lenght: 0
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F20 ACK Proxy B -> Bob's Phone
ACK sip:bob@bobphone.example.com SIP/2.0
Via: SIP/2.0/TLS proxyb.example.com:5061;branch=z9hG4bKbalouba.1
Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
Max-Forwards: 68
To: Bob <sip:bob@example.com>;tag=5551212
From: Alice <sip:alice@example.net>;tag=8675309
Call-ID: lzksjf8723k@sodk6587
CSeq: 1 ACK
Content-Lenght: 0
10. Conclusion
The restrictions described in this document have consequences on the
applicability of the SIPS URI scheme.
SIP [RFC3261] itself introduces some complications with using SIPS,
for example when using strict routing instead of loose routing. When
a SIPS URI is used in a Contact in a dialog initiating request and
Record-Route is not used, that SIPS URI may 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 may occur. In this case, using Record-Route so that the
requests are sent through proxies helps in making it work. Another
example of issues with strict routing is that even in a case where
the Contact is a SIPS URI, no Record-Route is used, and the far end
supports SIPS and TLS, it may still not be possible for the far end
to establish a TLS connection with the SIP originating end if the
certificate can not 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 even if the originating end is
not using a certificate that can be validated. In both cases,
[I-D.ietf-sip-outbound] and Record-Route may be used to solve the
problem.
TLS itself has a significant impact on how SIPS may 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 (like a phone or a personal
computer), and it's SIP server (proxy or registrar) acting as the TLS
server side. "Mutual TLS" (where both the client and the server side
provide their respective certificate) is typically used between SIP
servers (proxies, registrars), or statically configured devices such
as PSTN gateways or media servers. In the mutual TLS model, for two
entities to be able to establish a TLS connection requires that both
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side 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 must be possible for the entity establishing the connection (the
client) to validate the certificate from the server side. For
server-side authentication, [I-D.ietf-sip-outbound] is the
recommended approach. For mutual TLS, it means that one should be
very careful that the architecture of the network is such that
connections are made between entities that have access to each
other's credential. Record-Route [RFC3261] and Path [RFC3327] are
very useful in ensuring that previously established TLS connections
can be re-used. Other mechanism may also be used in certain
circumstances: for example, using root-certificates that are widely
recognized may allow for more easily creqated TLS connections.
The "last hop exception" introduces significant potential
vulnerabilities in SIP. Obviously, there is no garantee on the type
of security that will be used on that last hop as it will be
completely up to the target domain. Annother vulnerability is that
there is no way to ensure that the last hop will really be the last
hop: that hop could redirect or retarget to more hops. These hops
could even be outside of the original target domain, and it is
possible that the fact that it was retargeted by an entity that was
not secured through TLS may be undetectable.
11. Security Considerations
Most of this document can be considered to be security considerations
since it applies to the usage of the SIPS URI.
12. IANA Considerations
There are no IANA considerations.
13. IAB Considerations
There are no IAB considerations.
14. Acknowledgments
The author would like to thank Jon Peterson, Cullen Jennings, and
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Jonathan Rosenberg for their help. The author would like to thank
John Elwell, Paul Kyzivat, Eric Rescorla, Rifaat Shekh-Yusef, Peter
Reissner, Samir Srivastava, Tina Tsou, Brian Stucker and Patrick Ma
for their careful review and input.
15. References
15.1. 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.
[RFC3327] Willis, D. and B. Hoeneisen, "Session Initiation Protocol
(SIP) Extension Header Field for Registering Non-Adjacent
Contacts", RFC 3327, December 2002.
15.2. Informational References
[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.
[RFC3892] Sparks, R., "The Session Initiation Protocol (SIP)
Referred-By Mechanism", RFC 3892, 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
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(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.
[RFC4346] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.1", RFC 4346, April 2006.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[I-D.ietf-sip-outbound]
Jennings, C. and R. Mahy, "Managing Client Initiated
Connections in the Session Initiation Protocol (SIP)",
draft-ietf-sip-outbound-07 (work in progress),
January 2007.
[I-D.ietf-sip-gruu]
Rosenberg, J., "Obtaining and Using Globally Routable User
Agent (UA) URIs (GRUU) in the Session Initiation Protocol
(SIP)", draft-ietf-sip-gruu-11 (work in progress),
October 2006.
Appendix A. Future steps in specification
This section is a placeholder for a discussion of possible future
steps in specification, and the pros and cons of making such changes.
Protocol and normative changes to any specifications (such as RFC
3261) resulting from this discussion would be specified in futher
documents.
A.1. Deprecation of Last Hop Exception
[I-D.ietf-sip-outbound] provides an elegant and complete solution to
the problem that was intented to be be solved unsucessfully by the
last hop exception rule as described in Section 3. It is expected
that a further specification may deprecate the option of using the
last hop exception rule. It is therefore RECOMMENDED for current
implementation to avoid using the last hop exception and instead use
[I-D.ietf-sip-outbound].
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A.2. True-end-to-end SIP security
SIPS provides a mean to request that every SIP signalling hop be
secured using TLS. This is very different than end-to-end security
since it implies transitive trust. A true mechanism for end-to-end
SIP signalling is a potential area for future specification.
Appendix B. To-Be-Done
Need to look at Replaces [RFC3891], Join [RFC3911] and Target-Dialog.
For example, what if this header field is received in a request to a
SIPS URI but the dialog to which it relates has a SIP local target,
or vice-versa?
Third-party call control [RFC3725] may also have its own set of
issues to investigate.
REFER [RFC3515] and also [RFC3892] introduces its own set of issues
with sips:
o What if a UA with no support for TLS receives a SIPS URI in a
Refer-to header in a REFER request? Does it reject the REFER, or
accept REFER and send back a 416 in a NOTIFY (whouldn't work if
norefersub is used)?
o How should the UAC sending a REFER react if it receives a 416 in
response to the REFER?
o What if a UA with TLS support receives a SIP URI in a Refer-to
header? Is it allowed to "upgrade" to a SIPS URI? It is probably
a bad idea in most scenarios, unless it already knows that the
other ends supports TLS (and has a SIPS URI).
Handling of annomalies are not very well defined in [RFC3261]. What
if a UAS receives a SIP Contact replacing a SIPS contact in a target
refresh? Should the UAC tear down the dialog if it can not cope with
the unexpected response?
For GRUU, how should the UAC react if the returned GRUU is SIP but
the To was SIPS? How should the UAC react if the returned GRUU is
SIPS but the To was SIP?
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Author's Address
Francois Audet
Nortel Networks
4655 Great America Parkway
Santa Clara, CA 95054
US
Phone: +1 408 495 2456
Email: audet@nortel.com
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Full Copyright Statement
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contained in BCP 78, and except as set forth therein, the authors
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Audet Expires August 11, 2007 [Page 48]
