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Versions: (draft-audet-sip-sips-guidelines) 00 01 02 03 04 05 06 07 08 09 RFC 5630

SIP                                                             F. Audet
Internet-Draft                                           Nortel Networks
Intended status: Standards Track                          April 13, 2007
Expires: October 15, 2007


The use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)
                         draft-ietf-sip-sips-03

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   This Internet-Draft will expire on October 15, 2007.

Copyright Notice

   Copyright (C) The IETF Trust (2007).

Abstract

   This document provides clarifications and guidelines concerning the
   use of SIPS URI scheme in the Session Initiation Protocol (SIP).  It
   also makes normative changes to SIP.  This document also provides a
   discussion of possible future steps in specification.







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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Overview . . . . . . . . . . . . . . . . . . . . . . . . . . .  3
     3.1.  Meaning of SIPS  . . . . . . . . . . . . . . . . . . . . .  3
       3.1.1.  Scope of SIPS  . . . . . . . . . . . . . . . . . . . .  6
       3.1.2.  Using TLS with SIP instead of SIPS . . . . . . . . . .  6
     3.2.  Routing  . . . . . . . . . . . . . . . . . . . . . . . . .  7
       3.2.1.  Detection of hop-by-hop security . . . . . . . . . . .  9
       3.2.2.  Double Record Routing  . . . . . . . . . . . . . . . .  9
     3.3.  Usage of tls transport parameter and TLS Via parameter . . 11
   4.  Normative Requirements . . . . . . . . . . . . . . . . . . . . 11
     4.1.  General User Agent Behavior  . . . . . . . . . . . . . . . 11
       4.1.1.  Service routes . . . . . . . . . . . . . . . . . . . . 13
       4.1.2.  Registration . . . . . . . . . . . . . . . . . . . . . 13
       4.1.3.  SIPS in a dialog . . . . . . . . . . . . . . . . . . . 15
       4.1.4.  Derived dialogs and transactions . . . . . . . . . . . 16
       4.1.5.  Usage of tls transport parameter . . . . . . . . . . . 16
       4.1.6.  GRUU . . . . . . . . . . . . . . . . . . . . . . . . . 17
     4.2.  Proxy Behavior . . . . . . . . . . . . . . . . . . . . . . 18
     4.3.  Redirect Server Behavior . . . . . . . . . . . . . . . . . 19
   5.  Call Flows . . . . . . . . . . . . . . . . . . . . . . . . . . 20
     5.1.  Bob Registers His Contacts . . . . . . . . . . . . . . . . 21
     5.2.  Alice Calls Bob's SIPS AOR . . . . . . . . . . . . . . . . 26
     5.3.  Alice Calls Bob's SIP AOR using TCP  . . . . . . . . . . . 31
     5.4.  Alice Calls Bob's SIP AOR using TLS  . . . . . . . . . . . 41
   6.  Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . 48
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 49
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 49
     8.1.  SIP Option Tag . . . . . . . . . . . . . . . . . . . . . . 49
   9.  IAB Considerations . . . . . . . . . . . . . . . . . . . . . . 49
   10. Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 49
   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 50
     11.1. Normative References . . . . . . . . . . . . . . . . . . . 50
     11.2. Informational References . . . . . . . . . . . . . . . . . 50
   Appendix A.  Future steps in specification . . . . . . . . . . . . 51
     A.1.  Indication of validity of SIPS . . . . . . . . . . . . . . 51
     A.2.  True end-to-end encryption of SIP  . . . . . . . . . . . . 51
   Appendix B.  Bug Fixes for RFC 3261  . . . . . . . . . . . . . . . 51
   Appendix C.  Open issues . . . . . . . . . . . . . . . . . . . . . 53
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 53
   Intellectual Property and Copyright Statements . . . . . . . . . . 54








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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 implementers.

   This document provides clarifications and guidelines concerning the
   use of the SIPS URI scheme in the Session Initiation Protocol (SIP).
   It also makes normative changes to SIP.  This document also provides
   a discussion of possible future steps in specification.


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.  Overview

3.1.  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

   According to [RFC3261], if a@A is sending a request to sips:b@B, the
   following applies:
   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 not.  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 believed
   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 to
   not be transported over 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



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   satisfactory solution to this problem.  [I-D.ietf-sip-outbound] also
   solve the problem that if UA b is behind a NAT or Firewall, proxy B
   would not even be able to establish a TCP session in the first place.

   Furthermore, consider the problem of using SIPS inside a dialog.  If
   a@A sends a request to b@B using a SIPS Request-URI, then, according
   to [RFC3261]/8.1.1.8, "the contact header field 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
   in the request from b@B to B would be a SIPS URI.

   Because of all the problems caused by the last hop exception, this
   specification deprecates the last hop exception when forwarding a
   request to the last hop (see Section 4.2).  This will ensure that TLS
   is used on all hops all the way up to the remote target.

   The SIPS scheme implies transitive trust.  Obviously, there is
   nothing that prevents proxies from cheating (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 securely.  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 guarantees 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] provides an authenticated identity mechanism and a domain-
   to-domain integrity protection mechanism.

   Some have asked why is SIPS useful in a global open environment such
   as the Internet, if (when used in a Request-URI) it is not an
   absolute guarantee that the request will in fact be delivered over
   TLS on each hop?  Why is SIPS any different than just using TLS
   transport with SIP?  The difference is that using a SIPS URI in a
   Request-URI means that if you are instructing the network to use TLS
   over each hop, and if it is not possible, to reject the request:



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   i.e., that you would rather have the request fail than have the
   request delivered without TLS.  Just using TLS with a SIP Request-URI
   instead of a SIPS Request-URI implies a "best-effort" service: the
   request may or may not be delivered over TLS on each hop.

   Another common question is why not have a Proxy-Require and Require
   option tag that forces the use of TLS instead?  The answer is that it
   would only be functionally equivalent to using SIPS in a Request-URI.
   SIPS URIs however can be used in many other header fields: in Contact
   for registration, Contact in dialog-creating requests, Route, Record-
   Route, Path, From, To, Refer-To, Refer-By, etc.  This specification
   clarifies the significance of using SIPS URIs in these cases.  SIPS
   URIs can also be used in human-usable format (e.g., business cards,
   user interface, etc.).  SIPS URIs can even be used in other protocols
   that allow for including SIPS URIs (e.g., HTML).

3.1.1.  Scope of SIPS

   This document specifies that SIPS means that the SIP resource
   designated by the target SIPS URI is to be contacted securely, using
   TLS on each hop between the UAC and the remote UAS (as opposed to
   only to the proxy responsible for the target domain of the Request-
   URI).  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.

3.1.2.  Using TLS with SIP instead of SIPS

   Because a SIPS URI implies that requests sent to the resource
   identified by it be sent over each SIP hop over TLS, SIPS URIs are
   not suitable for "best-effort TLS": they are only suitable for "TLS-
   only" 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.  In fact,
   implementations SHOULD try to establish a TLS connection when using a
   SIP URI.

   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 header field, Route, etc.).
   However, when TLS is used, the Via header indicates 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



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   URIs.  The proxies may insert Record-Route using SIP URIs even if it
   uses TLS transport.

   Some user agents, redirect server and proxies may have local policies
   that enforce TLS on all connections, independently of if SIPS is used
   or not.

3.2.  Routing

   SIP and SIPS URIs that are identical except for the scheme itself
   (e.g., sip:alice@example.com and sips:alice@example.com) 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 securely".  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 either that proxies should arbitrarily
   "upgrade" SIP URIs to SIPS URIs when forwarding a request (See
   Section 4.2).  Rather, it means that when a resource is addressable
   with SIP, it will also be addressable with SIPS.

   For example, consider the case of a UA that has registered with SIPS
   contact header field.  If a UAC later addresses a request using a SIP
   Request-URI, the proxy will forward the request addressed to a SIP
   Request-URI to the endpoint, as illustrated by message F13 in
   Section 5.3 and in Section 5.4.  The proxy forwards the request to
   the UA using a SIP Request-URI and not the SIPS Request-URI used in
   registration (and it does this by substituting the sip scheme to the
   sips scheme in the registered contact header field 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 header field.  That SIPS contact header field would then
   force the other UA to use a SIPS contact header field in any mid-
   dialog request, including the ACK (which wouldn't be possible if that
   UA did not support SIPS).

   This specification mandates that a resource described by a SIPS
   Request-URI can not be "downgraded" to a SIP URI when a proxy is
   forwarding a request by changing the scheme, or by sending the
   associated request over a non secure link.  See Section 4.2.

   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



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   [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 header field URIs in
   REGISTER requests.  The key point is that this statement applies to
   the contact header field bindings in a registration: it is the
   association of the contact header field 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 header
   field (e.g., sips:bob@bobphone.example.com), the registrar 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@example.com, Bob's proxy will route it to Bob at Request-URI
   sip:bob@bobphone.example.com.  The proxy should attempt to transport
   the request over TLS if a TLS connection can be established even if a
   SIP URI is used.  Indeed, some proxies may even have local policies
   of always using TLS.  Furthermore, if Bob wants to ensure that every
   request delivered to it be always transported over TLS, Bob can use
   [I-D.ietf-sip-outbound] when registering.

   However, if Bob had registered instead with a SIP contact header
   field instead of a SIPS contact header field (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 header field for Bob, and "downgrades" from SIPS to SIP
   are not allowed.

   See Section 5 for illustrative call flows.

   Since upgrading from SIP to SIPS is allowed in 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 (Unsupported URI scheme), because the UAS only
   supports the SIP scheme.  When 416 (Unsupported URI scheme) is
   received, the request may be re-attempted with a SIP URI, but the
   user should be informed.  While guessing a SIPS AOR from a known SIP
   AOR and using it to initiate a request is a valid thing to do, doing
   the opposite (i.e., guessing a SIP AOR from a SIPS AOR and using it)
   is not a valid thing to do as it would be a security downgrade.

   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



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   SIPS URI with a contact header field containing a SIP URI.
   [RFC3261]/8.1.3.4 states 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 would not be able
   recurse on the contact header field, and instead would either forward
   the 3XX to the UAC or reject the request.

3.2.1.  Detection of hop-by-hop security

   The presence of a SIPS Request-URI does not necessarily indicate that
   the request was sent securely on each hop.  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.  Additionally, the History-Info header
   [RFC4244] could be inspected for detecting retargeting between SIP
   and SIPS.

   It should be restated that all the checking may be circumvented by
   any proxies or B2BUAs on the path that does not follow the rules and
   recommendations of this specification and of [RFC3261].

   Proxies can have their own policies regarding routing of requests to
   SIP or SIPS URIs.  For example, some proxies in some environment may
   be configured to only route SIPS.  Some proxies may be configured to
   detect non-compliances and reject un-secure requests.  For example,
   proxies could inspect Request-URIs, Path, Record-Route, To, From,
   contact header field 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, a
   preferable 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).

3.2.2.  Double Record Routing

   While proxies conforming to this specification do not forward or
   retarget from SIP to SIPS and vice-versa, it is possible that proxies
   that conform to [RFC3261] but not to this specification may do so.



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   The use case for a proxy to forward a request from SIPS to SIP was
   the "last hop exception" downgrade described in Section 1.

   This section explains how such a proxy would be able to use "double
   record route" in order to forward or retarget a request from SIP to
   SIPS or from SIPS to SIP.  This section is included for completeness,
   to describe how to achieve backward compatibility.

   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.  [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.
   It is preferable to use the procedures of [RFC3608], and instead of
   following the procedure in [RFC3261], proxies that are inserting
   Record-Route or Path header field URIs would 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 scheme 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.

   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-URI 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:






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       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].

3.3.  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.  The reference in section 26.2.1 is
   an error in [RFC3261] (see Appendix B).  However, the parameter needs
   to remain in the ABNF for backward compatibility in order for parsers
   to be able to process the parameter correctly.

   For Via headers, the following transport protocol are defined in
   [RFC3261]: "UDP", "TCP", "TLS", "SCTP", and in [RFC4168]: "TLS-SCTP".


4.  Normative Requirements

   This section describes all the normative requirements defined by this
   specification.  The justification for the [RFC2119] language is
   provided by the previous sections of this specification.

4.1.  General User Agent Behavior

   When presented with a SIPS URI, a UAC or UAS MUST NOT map it to a SIP
   URI.  For example, if a directory entry includes a SIPS AOR, the UAC



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   must not send requests to that AOR using a SIP Request-URI.
   Similarly, if a user reads a business card with a SIPS URI, he should
   not infer a SIP URI (unfortunately, we can not prevent people from
   being foolish).  If a 3XX response includes a SIPS contact header
   field, the UAC MUST NOT replace it with a SIP Request-URI (e.g., by
   replacing the SIPS scheme with a SIP scheme) when sending a request
   as a result of the redirection.

   A UAC that conforms to this specification MUST include a sips option-
   tag in Supported or Require header field when sending a request.  The
   Supported header is useful for a "backward compatible mode" where the
   UAS may not support this specification, but still support SIPS as per
   [RFC3261] (note however that since the usage of SIPS was greatly
   underspecified in [RFC3261], there is no guarantee that it will
   actually work).  The Require header field is useful for a mode where
   the expectation is that this specification be supported throughout
   the network (for example, to enforce the deprecation of the last hop
   exception).  If the Request-URI is a SIP URI, a sips option-tag in a
   Require header field MUST NOT be used, and a Supported header field
   MUST be used instead.  If a UAS receives a request with the sips
   option-tag in a Supported or Require header field and it accepts the
   registration, it MUST include the sips option-tag in Supported or
   Require header in a 200 (OK) response.  If the UAS does not support
   the sips option-tag, it will reject a request with a sips option-tag
   in a Require header field, and it will no include the sips option-tag
   in a 200 (OK) sent in response to a request with a sips option-tag in
   a Supported header.  If a UAC sent a request with a sips option-tag
   in a Supported header, and the 200 (OK) did not include a sips
   option-tag in a Require or Supported header, it will know that the
   procedures of this specification are not supported by the UAS.  The
   UAC MUST include the sips option-tag in a Proxy-Require header in a
   request if it wants to ensure that this specification is supported by
   all proxies along the path (this is not suitable for a "backward
   compatible mode").

   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 does not support SIPS, "it SHOULD reject the request with a
   416 (Unsupported URI scheme) response" as described in [RFC3261]/
   8.2.2.1.  Upon receiving a 416, a UAC MUST NOT re-attempt the request
   by automatically replacing the SIPS scheme with a SIP scheme as



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   described in [RFC3261]/8.1.3.5 as it would be a security
   vulnerability.  If the UAC does re-attempt the call with a SIP URI,
   it SHOULD get a confirmation from the user to authorize re-initiating
   the session with a SIP Requst-URI instead of a SIPS Request-URI.

   If a UAS does not wish to be contacted with a SIP URI, it MAY reject
   a request to a SIP Request-URI with response code 404 (Not Found), or
   it MAY redirect the request to a SIPS URI with a 3XX response.  A 3XX
   response has the advantage that it provides some indication to the
   UAC on why the request was rejected, i.e., that the session SHOULD be
   tried again to the SIPS Contact header field.  Using a 404 (Not
   Found) provides no useful indication on why the request is rejected.
   Upon receiving a 3XX response with a SIPS Contact header field, the
   UAC SHOULD automatically re-initiate the request using a SIPS
   Request-URI (i.e., it does not need to get a confirmation from the
   user to autorize re-initiating the session with a SIPS Request-URI
   instead of a SIP Request-URI).

4.1.1.  Service routes

   If a UA registers with a SIPS contact header field, the registrar
   returning a service route MUST return a service route consisting of
   SIP URIs if the intent of the registrar is to allow both SIP and SIPS
   to be used in requests sent by that client.  If a UA registers with a
   SIPS contact header field, the registrar returning a service route
   MUST return a service route consisting of SIPS URIs if the intent of
   the registrar is to allow only SIPS URIs to be used in requests sent
   by that UA.  It is the responsibility of the UAC to use a Route
   header consisting of all SIPS URIs when using a SIPS Request-URI and
   contact header field.  Specifically, 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.

4.1.2.  Registration

   This section describes the registration procedures of SIPS versus SIP
   contact header field that follows from the discussion in Section 3.2.

   The UAC 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.  The registrar MUST consider AORs
   that are identical except for one having the SIP scheme and the other
   having the SIPS scheme to be equivalent.

   If a UA wishes to be reachable with a SIPS URI, it MUST register with



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   a SIPS contact header field.  Requests addressed to that UA's AOR
   using either a SIP or SIPS Request-URI will be routed to that UA.
   Note that this includes UA that support both SIP and SIPS.

   If a UA does not wish to be reached with a SIPS URI, it MUST register
   with a SIP contact header field.  The UA MUST NOT include a sips
   option-tag in that case.

   Because registering with a SIPS contact header field implies a
   binding to both a SIPS Contact and a corresponding SIP Contact, a UA
   MUST NOT include both the SIPS and the SIP version of the same
   contact header field in a REGISTER: it MUST only use the SIPS version
   in this case.  Similarly, a UA SHOULD NOT register both a SIP contact
   header field and a SIPS contact header field in separate
   registrations as the SIP contact header field would be superfluous.
   Note however that a UA could register first with a SIP contact header
   field (meaning it does not support SIPS), and later register with a
   SIPS contact header field (meaning it now supports SIPS).

   If a UA wants to ensure that no requests are delivered without using
   TLS on that last hop, the UA MUST use [I-D.ietf-sip-outbound].

   If all the contact header fields in a REGISTER are SIPS, a SIPS AOR
   MUST be used by the UAC in the REGISTER.  If at least one of the
   contact header fields 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 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).  These
   are purely 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" would address the Registrar using a SIPS Request-URI
   over TLS, would register with a SIPS contact header field, and would
   reject requests addressed to a SIP Request-URI with 403 (Forbidden).
   A UA with a policy of "best-effort SIPS" would address the Registrar
   using a SIPS Request-URI over TLS, would register with a SIPS contact
   header field, and would accept requests addressed to either SIP or
   SIPS Request-URIs.  A UA with a policy of "No SIPS" would address the
   Registrar using a SIP Request-URI, could use TLS or not, would
   register with a SIP AOR and a SIP contact header field, and would
   accept requests addressed to a SIP Request-URI.

   A registrar MUST only accept a binding to a SIPS contact header field
   if all the appropriate URIs are of the SIPS scheme, otherwise there
   could be an inadvertent binding of a secure resource (SIPS) to an
   unsecured one (SIP).  This includes the Request-URI, the Contacts and



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   all the Path headers, but does not include the From and To headers.
   If the URIs are not of the proper SIPS scheme, the registrar MUST
   reject the REGISTER with a 403 (Forbidden).

   The usage of the "transport" URI parameter in contact header fields
   in registration is of dubious usefulness.  The assumption is that a
   UAC would 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 contact header fields
   (e.g, udp, TCP, sctp, tls 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 contact header fields in REGISTER.

   For backward compatibility, a registrar should accept a REGISTER
   message with a transport URI parameter in the contact header field.
   It is recommended that a registrar ignores that parameter, i.e., that
   it will not influence routing.

4.1.3.  SIPS in a dialog

   If the Request-URI in a request that initiates a dialog is a SIP URI,
   then the UAC needs to be careful about what to use in the contact
   header field (in case Record-Route is not used for this hop).  If the
   contact header field was a SIPS URI, it would mean that it would only
   accept mid-dialog requests that are sent over secure transport on
   each hop.  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.  If the top Route header field
   does not contain a SIPS URI, the contact header field MUST 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 and UAS MUST
   include a contact header field with a SIPS URI if the original
   request used a SIPS Request-URI.





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4.1.4.  Derived dialogs and transactions

   Sessions, dialogs and transactions may be "derived" from existing
   ones.  A good example of a derived dialog is one that was established
   as a result of using the REFER method [RFC3515].

   As a general principle, derived dialogs and transactions MUST NOT
   result in an effective downgrading of SIPS to SIP, without the
   explicit authorization of the entities involved.

   For example, when a REFER request is used to perform a call transfer,
   it results in an existing dialog being terminated and another one
   being created based on the Refer-to URI.  If that initial dialog was
   established using SIPS, then the new one MUST NOT be established
   using SIP, unless there is an explicit authorization given by the
   recipient of REFER.  This could be a warning provided to the user.
   Having such a warning could be useful for example for a secure
   directory service application, resulting in being routed to a UA that
   does not support SIPS.  If the proper treatment is to reject the
   REFER, for example because warnings are impractical or impossible
   with very simple phones, it could be rejected with error response 404
   (Not Found).

   Note that a REFER may also be used for referring to resources that do
   not result in dialogs being created.  In fact, a REFER may be used to
   point to resources that are of a different type than the original one
   (i.e., not SIP or SIPS).  Please see [RFC3515]/5.2 for security
   considerations related to this.

   Other examples of derived dialogs and transactions include the use of
   Third-Party Call Control [RFC3725], the Replaces header [RFC3891],
   the Join header [RFC3911].  Again, the general principle is that
   these mechanism SHOULD NOT result in an effective downgrading of SIPS
   to SIP, without the proper authorization.

4.1.5.  Usage of tls transport parameter

   The "transport=tls" parameter MUST NOT be used by UAs.  However, for
   backward compatibility, if a "transport=tls" parameter is received by
   a UA, it should be interpreted as per the following guidelines:

   o  [RFC3261]/16.7 states that in a Record-Route, "The URI SHOULD NOT
      contain the transport parameter unless the proxy has knowledge
      (such as in a private network) that the next downstream element
      that will be in the path of subsequent requests 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



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      NOT be used in Route, Record-Route or Path headers, and MUST be
      ignored.
   o  In a contact header field in a dialog, it could 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 presence of the
      transport parameter.  It is RECOMMENDED that the "transport=tls"
      parameter in a Contact header field in a dialog be ignored by the
      UAS.
   o  In a contact header field 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 by the registrar in a Contact header field in a REGISTER.
   o  In a Request-URI, the transport parameter is problematic.  On the
      last hop, it is useless because the transport is evident.  A UAS
      MUST ignore the "transport=tls" parameter in a Request-URI.
   o  In a contact header field 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.

4.1.6.  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 Contact header in the REGISTER
   request contains a SIP URI, the SIP version of the GRUU is returned.
   If the Contact header filed in the REGISTER request contains a SIPS
   URI, the SIPS version of the GRUU is returned.

   If the wrong scheme is received in the GRUU (which would be an error
   in the registrar), the UAC SHOULD treat it as if the proper scheme
   was used (i.e., it SHOULD replace the scheme with the proper scheme
   before using the GRUU).






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4.2.  Proxy Behavior

   Proxies that conform to this specification MUST NOT use the last hop
   exception when forwarding or retargeting a request to the last hop.
   Specifically, when a proxy receives a request with a SIPS Request-
   URI, the proxy MUST forward or retarget the request to a SIPS
   Request-URI.  If the target UAS had registered previously using a SIP
   Contact header field instead of a SIPS Contact header field, the
   proxy MUST NOT forward the request to the URI indicated in the SIPS
   Contact header field.  If the proxy needs to reject the request for
   that reason, it MUST reject it with a 404 (Not Found).

   Proxies SHOULD transport requests using a SIP URI over TLS when it is
   possible to set-up a TLS connection, or re-use an existing one.
   [I-D.ietf-sip-outbound] for example, allows for re-using an existing
   TLS connection.  Some proxies MAY have policies that prohibits
   sending any request over anything but TLS.

   When a proxy receives a request with a SIP Request-URI, the proxy
   MUST NOT forward the request to a SIPS Request-URI.  If the target
   UAS had registered previously using a SIPS Contact header field, and
   the proxy decides to forward the request, it MUST substitute the SIP
   scheme to the SIPS scheme of the URI found in the registered Contact
   header field binding and use it in the Request-URI of the forwarded
   request, and those proxies MUST use TLS to forward the request to the
   UAS.  Some proxies MAY have a policy of not forwarding at all
   requests using a non-SIPS Request-URI if the UAS had registed using a
   SIPS Contact header fields.  If the proxy elects to reject the
   request because it has such a policy or because it is not capable of
   establishing a TLS connection, it MAY reject it with a 404 (Not
   found) or it MAY redirect it to a SIP Request-URI with a 3XX
   response.  A 3XX response has the advantage that it provides some
   indication to the UAC on why the request was rejected, i.e., that the
   session SHOULD be tried again to the SIPS Contact header field.
   Using a 404 (Not Found) provides no useful indication on why the
   request is rejected.

   Proxies can use the sips option-tag in Supported, Require and Proxy-
   Require header fields to detect UAs that do not conform to this
   specification.  This is particularly useful for backward
   compatibility with legacy implementations that included support for
   SIPS before the publication of this specification.  It should be
   noted that because the usage of the SIPS URI scheme was
   underspecified in [RFC3261], proxies may have to do fairly complex
   implementation-specific non-compliant procedures to handle legacy
   implementations.

   It is RECOMMENDED to use an outbound proxy as per the procedures



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   defined in [I-D.ietf-sip-outbound] for supporting UACs that can not
   provide a certificate for establishing a TLS connection (i.e., when
   server-side authentication is used).

   When a proxy sends a request using a SIPS Request-URI and receives a
   3XX response with a SIP Contact header field, it MUST NOT recurse on
   the Contact header field.  The Proxy SHOULD forward the 3XX to the
   UAC instead of recursing, in order to allow for the UAC to take the
   appropriate action.  The proxy MAY instead reject the request with a
   404 (Not found) if it is not its policy to allow redirection to be
   done by the UA and consequently, the user will not receive any
   indication of why the request was rejected.

   When a proxy sends a request using a SIP Request-URI and receives a
   3XX response with a SIPS Contact header field, it MUST NOT recurse on
   the Contact header field.  The Proxy SHOULD forward the 3XX to the
   UAC instead of recursing, in order to allow for the UAC to take the
   appropriate action.  The proxy MAY instead map the request with a 416
   (Unsupported URI Scheme) if it is not its policy to allow redirection
   to be done by the UA.

   The "transport=tls" parameter must not be used by proxies.  However,
   for backward compatibility, if a "transport=tls" parameter is
   received by a proxy, it should be interpreted as per the following
   guidelines:

   o  In a Request-URI, the transport parameter is problematic.  Before
      being resolved to the last hop (with loose routing), it is not
      clear what it would mean to the proxy.  A proxy MUST ignore the
      "transport=tls" parameter in a Request-URI.

4.3.  Redirect Server Behavior

   Using a Redirect Server instead of a Proxy, with TLS has some
   limitations that has to be taken into account.  Since there no pre-
   established connection (such as with [I-D.ietf-sip-outbound]) between
   the Proxy and the UAS, it is only appropriate for scenarios where
   inbound connections are allowed.  For example, it could be used in a
   server to server environment (redirect server or proxy server) where
   mutual TLS is used, and where there is no NAT traversal issues.  A
   redirect server would not be usable if server-side authentication is
   used or if there is a NAT between the server and the UAS.

   When a redirect server receives a request with a SIP Request-URI, the
   redirect server MAY redirect with a 3XX response to either a SIP or a
   SIPS Contact header field.  If the target UAS had registered
   previously using a SIPS Contact header field, the redirect server
   SHOULD return a SIPS Contact header field to "upgrade" to SIPS if it



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   is in an environment where TLS is usable (as described in the
   previous paragraph).  If the target UAS had registered previously
   using a SIP Contact header field, the redirect server MUST return a
   SIP Contact header field in a 3XX response if it redirects the
   request.

   When a redirect server receives a request with a SIPS Request-URI,
   the redirect server MAY redirect with a 3XX response to either a SIP
   or a SIPS Contact header field.  If the target UAS had registered
   previously using a SIPS Contact header field, the redirect server
   SHOULD return a SIPS Contact header field to "upgrade" to SIPS if it
   is in an environment where TLS is usable (as described in the
   previous paragraph).  If the target UAS had registered previously
   using a SIP Contact header field, the redirect server MUST return a
   SIP Contact header field in a 3XX response if it chooses to redirect;
   otherwise it may reject the request with a 416 (Unsupported URI
   Scheme).


5.  Call Flows

   The following diagram illustrates the topology used for the examples
   in this section:


                         |-----------|
                         | Registrar |
                         |-----------|
                               |
                               |
                         |-----------|          |-----------|
                         |  Outbound |__________|  Outbound |
                         |  Proxy B  |          |  Proxy A  |
                         |-----------|          |-----------|
                          /        |                  |
                         /         |                  |
                        /          |                  |
                 ______            |                  |
                |      |         _____              _____
                |______|        O / \ O            O / \ O
               /_______/         /___\              /___\

               bob@bobppc     bob@bobphone          Alice


                                 Topology

   In the following examples, Bob has two clients, one is a SIP PC



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   client running on his computer, and the other one is a SIP Phone.
   The PC client does not support SIPS and consequently only registers
   with a SIP contact header field.  The SIP phone however does support
   SIPS and TLS, and consequently registers with a SIPS contact header
   field.  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.  The registration process call flow is illustrated in
   Section 5.1.

   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 Outbound 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 Outbound Proxy B.

   The following subsections illustrates registration and three
   examples.  In the first example (Section 5.2), Alice calls Bob using
   Bob's SIPS URI.  In the second example (Section 5.3), Alice calls
   Bob's SIP AOR using TCP transport.  In the third example
   (Section 5.4), Alice calls Bob's SIP AOR using TLS transport.

5.1.  Bob Registers His Contacts

   This call flow illustrates the registration process by which Bob's
   device registers.  His PC client (Bob@bobpc) registers with a SIP
   scheme and his SIP Phone (Bob@phone) registers with a SIPS scheme.



















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                                    Outbound
                 Bob@bobpc          Proxy B      Registrar
                  |                   |               |
                  |    REGISTER F1    |               |
                  |------------------>|  REGISTER F2  |
                  |                   |-------------->|
                  |                   |    200 F3     |
                  |      200 F4       |<--------------|
                  |------------------>|               |
                  |                   |               |
                  |   Bob@bobphone    |               |
                  |      |            |               |
                  |      |REGISTER F5 |               |
                  |      |----------->|  REGISTER F6  |
                  |      |            |-------------->|
                  |      |            |    200 F7     |
                  |      |   200 F8   |<--------------|
                  |      |----------->|               |
                  |      |            |               |


                        Bob Registers His Contacts

   Message details


     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









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     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


     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















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     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


     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: sips, 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

















Audet                   Expires October 15, 2007               [Page 24]


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     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: sips, 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


     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: sips, 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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     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: sips, 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

5.2.  Alice Calls Bob's SIPS AOR

   Bob's registration has already occurred as per Section 5.1.

   In this first example, Alice calls Bob's SIPS AOR
   (sips:bob@example.com).  Proxy B consults the binding in the
   registration database, and finds the 2 contact header field 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 header field, and therefore the
   request is only sent to sips:bob@bobphone.example.com.  Proxy B
   inserts itself in the Record-Route.  Bob answers at
   sips:bob@bobphone.example.com.





















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                               Outbound       Outbound
          Bob@bobpc            Proxy B        Proxy A       Alice
           |                     |              |              |
           |   Bob@bobphone      |              |  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


     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
     Supported: sips
     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}












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     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
     Supported: sips
     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}


     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











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     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
     Supported: sips
     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
     Supported: sips
     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
     Supported: sips
     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-Length: 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-Length: 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-Length: 0

5.3.  Alice Calls Bob's SIP AOR using TCP

   Bob's registration has already occurred as per Section 5.1.

   In the second example, Alice calls Bob's SIP AOR 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



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   contact header field bindings.  Alice had addressed Bob with a SIP
   Request-URI (sip:bob@example.com), so Proxy B determines that the
   calls needs to be routed both to the SIP contact header field and the
   SIPS contact header field, and therefore the request is forked sent
   to sip:bob@boppc.example.com and sip:bob@bobphone.example.com.  Note
   that the proxy substituted the SIP scheme to the SIPS scheme for
   bob@bobphone.example.com.  Outbound 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        Proxy A       Alice
            |                   |              |             |
            |                   |              |  INVITE F9  |
            |                   |  INVITE F11  |<------------|
            |     INVITE F13'   |<-------------|   100 F10   |
            |<------------------|   100 F12    |------------>|
            |      100 F14'     |------------->|             |
            |------------------>|              |             |
            |      180 F15'     |              |             |
            |------------------>|   180 F16'   |             |
            |                   |------------->|   180 F17'  |
            |   Bob@bobphone    |              |------------>|
            |      |            |              |             |
            |      | 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

   Message details












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     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
     Supported: sips
     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


     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
     Supported: sips
     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}






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     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


     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
     Supported: sips
     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








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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 October 15, 2007               [Page 36]


Internet-Draft                    SIPS                        April 2007


     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
     Supported: sips
     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 October 15, 2007               [Page 37]


Internet-Draft                    SIPS                        April 2007


     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
     Supported: sips
     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
     Supported: sips
     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 October 15, 2007               [Page 38]


Internet-Draft                    SIPS                        April 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
     Supported: sips
     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-Length: 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-Length: 0







Audet                   Expires October 15, 2007               [Page 39]


Internet-Draft                    SIPS                        April 2007


     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-Length: 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-Length: 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-Length: 0
















Audet                   Expires October 15, 2007               [Page 40]


Internet-Draft                    SIPS                        April 2007


     F22' 487 (INVITE) Proxy B -> Bob's PC Client

     SIP/2.0 487 Request Terminated
     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

5.4.  Alice Calls Bob's SIP AOR using TLS

   Bob's registration has already occurred as per Section 5.1.

   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 contact header fields, To and
   From however would also normally indicate a SIP URI.

   The call flow would be exactly as per the second example
   (Section 5.3).

   Messages F20'-F22' are identical to the ones in Section 5.3.  The
   other messages are as follows.
















Audet                   Expires October 15, 2007               [Page 41]


Internet-Draft                    SIPS                        April 2007


     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
     Supported: sips
     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
     Supported: sips
     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 October 15, 2007               [Page 42]


Internet-Draft                    SIPS                        April 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
     Supported: sips
     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 October 15, 2007               [Page 43]


Internet-Draft                    SIPS                        April 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 October 15, 2007               [Page 44]


Internet-Draft                    SIPS                        April 2007


     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
     Supported: sips
     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 October 15, 2007               [Page 45]


Internet-Draft                    SIPS                        April 2007


     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
     Supported: sips
     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
     Supported: sips
     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 October 15, 2007               [Page 46]


Internet-Draft                    SIPS                        April 2007


     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
     Supported: sips
     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-Length: 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-Length: 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-Length: 0


6.  Conclusion

   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 header field 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 may help in making it
   work.  Another example of issues with strict routing 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 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 if the
   originating end is not using a certificate that can be validated.

   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 (e.g., 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, 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



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   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 created TLS connections.

   The "last hop exception" introduces significant potential
   vulnerabilities in SIP and therefore has been deprecated by this
   specification.


7.  Security Considerations

   Most of this document can be considered to be security considerations
   since it applies to the usage of the SIPS URI.


8.  IANA Considerations

8.1.  SIP Option Tag

   This specification registers one new SIP option tag, as per the
   guidelines in section 27.1 of [RFC3261].

   Name:  sips
   Description:  This option-tag is used to identify UACs, UASs and
      Registrars that support the procedures of this specification.


9.  IAB Considerations

   There are no IAB considerations.


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 and Dean Willis
   for their careful review and input.



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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.

   [RFC3327]  Willis, D. and B. Hoeneisen, "Session Initiation Protocol
              (SIP) Extension Header Field for Registering Non-Adjacent
              Contacts", RFC 3327, December 2002.

11.2.  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.

   [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,



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              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-08 (work in progress), March 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-13 (work in progress),
              April 2007.


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 further
   documents.

A.1.  Indication of validity of SIPS

   Since the presence of a SIPS URI in a Request-URI in an incoming
   request currently does not guarantee that SIPS and TLS was indeed
   used on every hop along the path, it has been suggested that it would
   be useful to define a mechanism for a verifiable assertion that TLS
   and SIPS were used on every hop.

A.2.  True end-to-end encryption of SIP

   Another suggestion has been to define a mechanism to encrypt SIP end-
   to-end.  This would require either an peer-to-peer SIP model, or
   alternatively a mechanism that allows the encrypted SIP signalling to
   be tunnelled through proxies.


Appendix B.  Bug Fixes for RFC 3261

   The fifth paragraph of section 10.2.1 should be replaced by:



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      If the address-of-record in the To header field of a REGISTER
      request is a SIPS URI, then any Contact header field value in the
      request MUST also be SIPS URIs.

   In section 16.7 on p. 112 describing Record-Route, the second
   paragraph should be deleted.

   The last paragraph of section 19.1 needs to be 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 section 26.2.1 delete the following phrase in the 6th paragraph:

      ; "tls" (signifying TLS over TCP) can be specified as the desired
      transport protocol within a Via header field value or a SIP-URI"

   The second paragraph of section 26.2.2 needs to be 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,
      must be 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 needs to be replaced by the
   following:

      Actually using TLS on every segment of a request path entails that
      the terminating UAS must be reachable over TLS (by registering
      with a SIPS URI as a contact).  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 fourth paragraph of section 26.4.4 needs to be deleted.

   The last sentence of the fifth paragraph of section 26.4.5 needs to
   be reworded as follows:



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      (...)  S/MIME or, preferably, [RFC4474] may also be used (...)


Appendix C.  Open issues

   1.  Do we deprecate the "last hop retarget upgrade exception"?  The
       document currently assumes that it IS deprecated, as this is the
       preference of the author.
       *  Advantages of deprecating are that it greatly simplifies the
          specification as there is no need for Double Record Route
          anymore, or for UAS to have to look at both To header and
          Request-URI to know if the request was indeed sent over TLS on
          every hop.  It also makes it much more consistent with the
          rest of the spec since it has been deprecated for "downgrades"
          as well as for "forwarding upgrades".
       *  Disadvantage is that it would not allow for automatically
          retargeting to a more secure mechanism.
       *  A counter argument is that you can still use TLS transport for
          the next hop, and in fact, we would recommend using it when
          possible.  This would be particularly obvious to do when sip-
          outbound is used.  Furthemore, you could also use redirection
          instead of retargeting (i.e., 3XX) to sips.
   2.  Do we need the sips option tag?
       *  Advantages of the option tag are that it allows for backward
          compatibility with implementations that have implemented SIPS
          as per RFC 3261, but not as per this specification.
       *  Disadvantage is, well, one more option tag.


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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