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RFC 5630 - The Use of the SIPS URI Scheme in the Session Initiat


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Network Working Group                                           F. Audet
Request for Comments: 5630                                    Skype Labs
Updates: 3261, 3608                                         October 2009
Category: Standards Track

The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)

Abstract

   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.

Status of This Memo

   This document specifies an Internet standards track protocol for the
   Internet community, and requests discussion and suggestions for
   improvements.  Please refer to the current edition of the "Internet
   Official Protocol Standards" (STD 1) for the standardization state
   and status of this protocol.  Distribution of this memo is unlimited.

Copyright and License Notice

   Copyright (c) 2009 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
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   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the BSD License.

   This document may contain material from IETF Documents or IETF
   Contributions published or made publicly available before November
   10, 2008.  The person(s) controlling the copyright in some of this
   material may not have granted the IETF Trust the right to allow
   modifications of such material outside the IETF Standards Process.
   Without obtaining an adequate license from the person(s) controlling
   the copyright in such materials, this document may not be modified
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   not be created outside the IETF Standards Process, except to format
   it for publication as an RFC or to translate it into languages other
   than English.

Table of Contents

   1. Introduction ....................................................3
   2. Terminology .....................................................3
   3. Background ......................................................3
      3.1. Models for Using TLS in SIP ................................3
           3.1.1. Server-Provided Certificate .........................3
           3.1.2. Mutual Authentication ...............................4
           3.1.3. Using TLS with SIP Instead of SIPS ..................4
           3.1.4. Usage of the transport=tls URI Parameter and
                  the TLS Via Parameter ...............................5
      3.2. Detection of Hop-by-Hop Security ...........................6
      3.3. The Problems with the Meaning of SIPS in RFC 3261 ..........7
   4. Overview of Operations ..........................................9
      4.1. Routing ...................................................11
   5. Normative Requirements .........................................13
      5.1. General User Agent Behavior ...............................13
           5.1.1. UAC Behavior .......................................13
                  5.1.1.1. Registration ..............................14
                  5.1.1.2. SIPS in a Dialog ..........................15
                  5.1.1.3. Derived Dialogs and Transactions ..........15
                  5.1.1.4. GRUU ......................................16
           5.1.2. UAS Behavior .......................................17
      5.2. Registrar Behavior ........................................18
           5.2.1. GRUU ...............................................18
      5.3. Proxy Behavior ............................................18
      5.4. Redirect Server Behavior ..................................20
   6. Call Flows .....................................................21
      6.1. Bob Registers His Contacts ................................22
      6.2. Alice Calls Bob's SIPS AOR ................................27
      6.3. Alice Calls Bob's SIP AOR Using TCP .......................36
      6.4. Alice Calls Bob's SIP AOR Using TLS .......................50
   7. Further Considerations .........................................51
   8. Security Considerations ........................................52
   9. IANA Considerations ............................................52
   10. Acknowledgments ...............................................52
   11. References ....................................................53
      11.1. Normative References .....................................53
      11.2. Informative References ...................................53
   Appendix A.  Bug Fixes for RFC 3261  ..............................55

1.  Introduction

   The meaning and usage of the SIPS URI scheme and of Transport Layer
   Security (TLS) [RFC5246] are underspecified in SIP [RFC3261] and have
   been a 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 (including both [RFC3261] and
   [RFC3608].

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.  Background

3.1.  Models for Using TLS in SIP

   This section describes briefly the usage of TLS in SIP.

3.1.1.  Server-Provided Certificate

   In this model, only the TLS server provides a certificate during the
   TLS handshake.  This is applicable only between a user agent (UA) and
   a proxy, where the UA is the TLS client and the proxy is the TLS
   server, and hence the UA uses TLS to authenticate the proxy but the
   proxy does not use TLS to authenticate the UA.  If the proxy needs to
   authenticate the UA, this can be achieved by SIP HTTP digest
   authentication.  This directionality implies that the TLS connection
   always needs to be set up by the UA (e.g., during the registration
   phase).  Since SIP allows for requests in both directions (e.g., an
   incoming call), the UA is expected to keep the TLS connection alive,
   and that connection is expected to be reused for both incoming and
   outgoing requests.

   This solution of having the UA always initiate and keep alive the
   connection also solves the Network Address Translation (NAT) and
   firewall problem as it ensures that responses and further requests
   will always be deliverable on the existing connection.

   [RFC5626] provides the mechanism for initiating and maintaining
   outbound connections in a standard interoperable way.

3.1.2.  Mutual Authentication

   In this model, both the TLS client and the TLS server provide a
   certificate in the TLS handshake phase.  When used between a UA and a
   proxy (or between two UAs), this implies that a UA is in possession
   of a certificate.  When sending a SIP request when there is not
   already a suitable TLS connection in place, a user agent client (UAC)
   takes on the role of TLS client in establishing a new TLS connection.
   When establishing a TLS connection for receipt of a SIP request, a
   user agent server (UAS) takes on the role of TLS server.  Because in
   SIP a UA or a proxy acts both as UAC and UAS depending on if it is
   sending or receiving requests, the symmetrical nature of mutual TLS
   is very convenient.  This allows for TLS connections to be set up or
   torn down at will and does not rely on keeping the TLS connection
   alive for further requests.

   However, there are some significant limitations.

   The first obvious limitation is not with mutual authentication per
   se, but with the model where the underlying TCP connection can be
   established by either side, interchangeably, which is not possible in
   many environments.  For examples, NATs and firewalls will often allow
   TCP connections to be established in one direction only.  This
   includes most residential SIP deployments, for example.  Mutual
   authentication can be used in those environments, but only if the
   connection is always started by the same side, for example, by using
   [RFC5626] as described in Section 3.1.1.  Having to rely on [RFC5626]
   in this case negates many of the advantages of mutual authentication.

   The second significant limitation is that mutual authentication
   requires both sides to exchange a certificate.  This has proven to be
   impractical in many environments, in particular for SIP UAs, because
   of the difficulties of setting up a certificate infrastructure for a
   wide population of users.

   For these reasons, mutual authentication is mostly used in server-to-
   server communications (e.g., between SIP proxies, or between proxies
   and gateways or media servers), and in environments where using
   certificates on both sides is possible (e.g., high-security devices
   used within an enterprise).

3.1.3.  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 26.2.2 of [RFC3261].

      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 send the request over TLS.

   Using SIP over TLS is very simple.  A UA opens a TLS connection and
   uses SIP URIs instead of SIPS URIs for all the header fields in a SIP
   message (From, To, Request-URI, Contact header field, Route, etc.).
   When TLS is used, the Via header field indicates TLS.

   [RFC3261], Section 26.3.2.1, states:

      When a UA comes online and registers with its local administrative
      domain, it SHOULD establish a TLS connection with its registrar
      (...).  Once the registration has been accepted by the registrar,
      the UA SHOULD leave this TLS connection open provided that the
      registrar also acts as the proxy server to which requests are sent
      for users in this administrative domain.  The existing TLS
      connection will be reused to deliver incoming requests to the UA
      that had just completed registration.

   [RFC5626] describes how to establish and maintain a TLS connection in
   environments where it can only be initiated by the UA.

   Similarly, proxies can forward requests using TLS if they can open a
   TLS connection, even if the route set used SIP URIs instead of SIPS
   URIs.  The proxies can insert Record-Route header fields using SIP
   URIs even if it uses TLS transport.  [RFC3261], Section 26.3.2.2,
   explains how interdomain requests can use TLS.

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

3.1.4.  Usage of the transport=tls URI Parameter and the TLS Via
        Parameter

   [RFC3261], Section 26.2.2 deprecated the "transport=tls" URI
   transport parameter in SIPS or SIP URIs:

      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.

   The "tls" parameter has not been eliminated from the ABNF in
   [RFC3261], Section 25, since the parameter needs to remain in the
   ABNF for backward compatibility in order for parsers to be able to
   process the parameter correctly.  The transport=tls parameter has
   never been defined in an RFC, but only in some of the Internet drafts
   between [RFC2543] and [RFC3261].

   This specification does not make use of the transport=tls parameter.

   The reinstatement of the transport=tls parameter, or an alternative
   mechanism for indicating the use of the TLS on a single hop in a URI,
   is outside the scope of this specification.

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

3.2.  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
   SIPS was used for the entire request path to secure the request end-
   to-end?  Effectively, the UAS cannot know for sure.  However,
   [RFC3261], Section 26.4.4, recommends how a UAS can make some checks
   to validate the security.  Additionally, the History-Info header
   field [RFC4244] could be inspected for detecting retargeting from SIP
   and SIPS.  Retargeting from SIP to SIPS by a proxy is an issue
   because it can leave the receiver of the request with the impression
   that the request was delivered securely on each hop, while in fact,
   it was not.

   To emphasize, all the checking can be circumvented by any proxies or
   back-to-back user agents (B2BUAs) on the path that do 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 environments can
   be configured to only route SIPS URIs.  Some proxies can be
   configured to detect non-compliances and reject unsecure requests.
   For example, proxies could inspect Request-URIs, Path, Record-Route,
   To, From, Contact header fields, and Via header fields to enforce
   SIPS.

   [RFC3261], Section 26.4.4, explains that S/MIME can 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], Section 26.4.4, this is meant to imply that [RFC3893]
   would be used to "tunnel" important header fields (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 header fields.  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, among other things, the Address of
   Record (AOR) of the sender (from the From header field) and the AOR
   of the original target (from the To header field).

3.3.  The Problems with the Meaning of SIPS in RFC 3261

   [RFC3261], Section 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.

        ..........................         ...........................
        .                        .         .                         .
        .              +-------+ .         . +-------+               .
        .              |       | .         . |       |               .
        .              | Proxy |-----TLS---- | Proxy |               .
        .              |   A   | .         . |  B    |               .
        .              |       | .         . |       |               .
        .            / +-------+ .         . +-------+ \             .
        .           /            .         .            \            .
        .          /             .         .             \           .
        .        TLS             .         .        Policy-based     .
        .        /               .         .               \         .
        .       /                .         .                \        .
        .      /                 .         .                 \       .
        .   +-------+            .         .              +-------+  .
        .   |       |            .         .              |       |  .
        .   | UAC a |            .         .              | UAS b |  .
        .   |       |            .         .              |       |  .
        .   +-------+            .         .              +-------+  .
        .             Domain A   .         .   Domain B              .
        ..........................         ...........................

                   SIP trapezoid with last-hop exception

   According to [RFC3261], if a@A is sending a request to sips:b@B, the
   following applies:

   o  TLS is required between UA a@A and Proxy A

   o  TLS is required between Proxy A and Proxy B

   o  TLS is required between Proxy B and UA b@B, depending on local
      policy.

   One can 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 [RFC5626].  At that time, it was
   recognized that in many practical deployments, Proxy B might 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 might 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, [RFC5626] has provided a more satisfactory
   solution to this problem.  [RFC5626] also solves 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], Section 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 consists of a SIPS URI, this implies that b@B is expected to
   understand SIPS in the first place, and is required to 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.

4.  Overview of Operations

   Because of all the problems described in Section 3.3, this
   specification deprecates the last-hop exception when forwarding a
   request to the last hop (see Section 5.3).  This will ensure that TLS
   is used on all hops all the way up to the remote target.

        ..........................         ...........................
        .                        .         .                         .
        .              +-------+ .         . +-------+               .
        .              |       | .         . |       |               .
        .              | Proxy |-----TLS---- | Proxy |               .
        .              |   A   | .         . |  B    |               .
        .              |       | .         . |       |               .
        .            / +-------+ .         . +-------+ \             .
        .           /            .         .            \            .
        .          /             .         .             \           .
        .        TLS             .         .             TLS         .
        .        /               .         .               \         .
        .       /                .         .                \        .
        .      /                 .         .                 \       .
        .   +-------+            .         .              +-------+  .
        .   |       |            .         .              |       |  .
        .   | UAC a |            .         .              | UAS b |  .
        .   |       |            .         .              |       |  .
        .   +-------+            .         .              +-------+  .
        .             Domain A   .         .   Domain B              .
        ..........................         ...........................

                 SIP trapezoid without last-hop exception

   The SIPS scheme implies transitive trust.  Obviously, there is
   nothing that prevents proxies from cheating (see [RFC3261], Section
   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 even a
   To header field) 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
   therefore the meaning of a SIPS URI is not to be oversold.  There is
   currently no mechanism to provide an indication of end-to-end
   security for SIP.  Other mechanisms can 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 from 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, 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 can but need
   not be delivered over TLS on each hop.

   Another common question is why not have a Proxy-Require and Require
   option tag forcing 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, Referred-By, etc.  SIPS URIs can
   also be used in human-usable format (e.g., business cards, user
   interface).  SIPS URIs can even be used in other protocols or
   document formats that allow for including SIPS URIs (e.g., HTML).

   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
   the SIP network, for example, to other networks such as the Public
   Switched Telephone Network (PSTN).

4.1.  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], Section
   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 cannot establish a secure connection to a client or
   another proxy.  This does not suggest either that proxies would
   arbitrarily "upgrade" SIP URIs to SIPS URIs when forwarding a request
   (see Section 5.3).  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 a
   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 UAS, as illustrated by message F13 in Sections
   6.3 and in 6.4.  The proxy forwards the request to the UA using a SIP
   Request-URI and not the SIPS Request-URI used in registration.  The
   proxy does this by replacing the SIPS scheme that was used in the

   registered Contact header field binding with a SIP scheme while
   leaving the rest of the URI as is, and then by using this new URI as
   the Request-URI.  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
   would not be possible if that UA did not support SIPS).

   This specification mandates that when a proxy is forwarding a
   request, a resource described by a SIPS Request-URI cannot be
   "downgraded" to a SIP URI by changing the scheme, or by sending the
   associated request over a nonsecure link.  If a request needs to be
   rejected because otherwise it would be a "downgrade", the request
   would be rejected with a 480 (Temporarily Unavailable) response
   (potentially with a Warning header with warn-code 380 "SIPS Not
   Allowed").  Similarly, this specification mandates that when a proxy
   is forwarding a request, a resource described by a SIP Request-URI
   cannot be "upgraded" to a SIPS URI by changing the scheme (otherwise
   it would be an "upgrade" only for that hop onwards rather than on all
   hops, and would therefore mislead the UAS).  If a request needs to be
   rejected because otherwise it would be a misleading "upgrade", the
   request would be rejected with a 480 (Temporarily Unavailable)
   response (potentially with a Warning header field with warn-code 381
   "SIPS Required").  See Section 5.3 for more details.

   For example, the sip:bob@example.com and sips:bob@example.com AORs
   refer to the same user "Bob" in the 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
   or sips:bob@example.com are treated the same way.  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 19.1.4 of [RFC3261] 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 whether or not the user is reachable with a SIPS URI.

   Consider this example: if Bob (AOR bob@example.com) registers with a
   SIPS Contact header field (e.g., sips:bob@bobphone.example.com), the
   registrar and the location service then know that Bob 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.

   If Bob wants to ensure that every request delivered to him always be
   transported over TLS, Bob can use [RFC5626] when registering.

   However, if Bob had registered 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 6 for illustrative call flows.

5.  Normative Requirements

   This section describes all the normative requirements defined by this
   specification.

5.1.  General User Agent Behavior

5.1.1.  UAC Behavior

   When presented with a SIPS URI, a UAC MUST NOT change it to a SIP
   URI.  For example, if a directory entry includes a SIPS AOR, the UAC
   is not expected to send requests to that AOR using a SIP Request-URI.
   Similarly, if a user reads a business card with a SIPS URI, it is not
   possible to infer a SIP URI.  If a 3XX response includes a SIPS
   Contact header field, the UAC does 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.

   As mandated by [RFC3261], Section 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".

   Upon receiving a 416 response or a 480 (Temporarily Unavailable)
   response with a Warning header with warn-code 380 "SIPS Not Allowed",
   a UAC MUST NOT re-attempt the request by automatically replacing the
   SIPS scheme with a SIP scheme as described in [RFC3261], Section
   8.1.3.5, as it would be a security vulnerability.  If the UAC does
   re-attempt the call with a SIP URI, the UAC SHOULD get a confirmation
   from the user to authorize re-initiating the session with a SIP
   Request-URI instead of a SIPS Request-URI.

   When the route set is not empty (e.g., when a service route [RFC3608]
   is returned by the registrar), it is the responsibility of the UAC to
   use a Route header field consisting of all SIPS URIs when using a
   SIPS Request-URI.  Specifically, if the route set included any SIP
   URI, the UAC MUST change the SIP URIs to SIPS URIs simply by changing
   the scheme from "sip" to "sips" before sending the request.  This
   allows for configuring or discovering one service route with all SIP
   URIs and allowing sending requests to both SIP and SIPS URIs.

   When the UAC is using a SIP Request-URI, if the route set is not
   empty and the topmost Route header field entry is a SIPS URI with the
   lr parameter, the UAC MUST send the request over TLS (using a SIP
   Request-URI).  If the route is not empty and the Route header field
   entry is a SIPS URI without the lr parameter, the UAC MUST send the
   request over TLS using a SIPS Request-URI corresponding to the
   topmost entry in the route set.

   To emphasize what is already defined in [RFC3261], UAs MUST NOT use
   the "transport=tls" parameter.

5.1.1.1.  Registration

   The UAC registers Contacts header fields to either a SIPS or a SIP
   AOR.

   If a UA wishes to be reachable with a SIPS URI, the UA MUST register
   with 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.
   This includes UAs that support both SIP and SIPS.  This specification
   does not provide any SIP-based mechanism for a UA to provision its
   proxy to only forward requests using a SIPS Request-URI.  A non-SIP
   mechanism such as a web interface could be used to provision such a
   preference.  A SIP mechanism for provisioning such a preference is
   outside the scope of this specification.

   If a UA does not wish to be reached with a SIPS URI, it MUST register
   with a SIP Contact header field.

   Because registering with a SIPS Contact header field implies a
   binding of both a SIPS Contact and a corresponding SIP Contact to the
   AOR, a UA MUST NOT include both the SIPS and the SIP versions of the
   same Contact header field in a REGISTER request; the UA 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.  If it does, the second registration replaces the first
   one (e.g., 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).  Similarly, if a
   UA registers first with a SIPS Contact header field and later
   registers with a SIP Contact header field, that SIP Contact header
   field replaces the SIPS Contact header field.

   [RFC5626] can be used by a UA if it wants to ensure that no requests
   are delivered to it without using the TLS connection it used when
   registering.

   If all the Contact header fields in a REGISTER request are SIPS, the
   UAC MUST use SIPS AORs in the From and To header fields in the
   REGISTER request.  If at least one of the Contact header fields is
   not SIPS (e.g., sip, mailto, tel, http, https), the UAC MUST use SIP
   AORs in the From and To header fields in the REGISTER request.

   To emphasize what is already defined in [RFC3261], UACs MUST NOT use
   the "transport=tls" parameter.

5.1.1.2.  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 the UAS would
   only accept mid-dialog requests that are sent over secure transport
   on each hop.  Since the Request-URI in this case is a SIP URI, it is
   quite possible that the UA sending a request to that URI might not be
   able to send requests to SIPS URIs.  If the top Route header field
   does not contain a SIPS URI, the UAC MUST use a SIP URI in the
   Contact header field, 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 [RFC5626]).

   When a target refresh occurs within a dialog (e.g., re-INVITE
   request, UPDATE request), the UAC MUST include a Contact header field
   with a SIPS URI if the original request used a SIPS Request-URI.

5.1.1.3.  Derived Dialogs and Transactions

   Sessions, dialogs, and transactions can 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 cannot
   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 UAC MUST NOT establish a new one
   using SIP, unless there is an explicit authorization given by the
   recipient of the REFER request.  This could be a warning provided to
   the user.  Having such a warning could be useful, for example, for a
   secure directory service application, to warn a user that a request
   may be routed to a UA that does not support SIPS.

   A REFER request can also be used for referring to resources that do
   not result in dialogs being created.  In fact, a REFER request can 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], Section
   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 field
   [RFC3891], and the Join header field [RFC3911].  Again, the general
   principle is that these mechanisms SHOULD NOT result in an effective
   downgrading of SIPS to SIP, without the proper authorization.

5.1.1.4.  GRUU

   When a Globally Routable User Agent URI (GRUU) [RFC5627] 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
   field in the REGISTER request contains a SIP URI, the SIP version of
   the GRUU is returned.  If the Contact header field 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).

5.1.2.  UAS Behavior

   When presented with a SIPS URI, a UAS MUST NOT change it to a SIP
   URI.

   As mandated by [RFC3261], Section 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 field in the response MUST
      be a SIPS URI.

   If a UAS does not wish to be reached with a SIPS URI but only with a
   SIP URI, the UAS MUST respond with a 480 (Temporarily Unavailable)
   response.  The UAS SHOULD include a Warning header with warn-code 380
   "SIPS Not Allowed".  [RFC3261], Section 8.2.2.1, states that UASs
   that do not support the SIPS URI scheme at all "SHOULD reject the
   request with a 416 (Unsupported URI scheme) response".

   If a UAS does not wish to be contacted with a SIP URI but instead by
   a SIPS URI, it MUST reject a request to a SIP Request-URI with a 480
   (Temporarily Unavailable) response.  The UAS SHOULD include a Warning
   header with warn-code 381 "SIPS Required".

   It is a matter of local policy for a UAS 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 the UAS would reject
   requests using the SIP scheme with a 480 (Temporarily Unavailable)
   response with a Warning header with warn-code 381 "SIPS Required".  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 the UAS 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 the
   UAS would accept requests addressed to a SIP Request-URI.

   If a UAS needs to reject a request because the URIs are used
   inconsistently (e.g., the Request-URI is a SIPS URI, but the Contact
   header field is a SIP URI), the UAS MUST reject the request with a
   400 (Bad Request) response.

   When a target refresh occurs within a dialog (e.g., re-INVITE
   request, UPDATE request), the UAS MUST include a Contact header field
   with a SIPS URI if the original request used a SIPS Request-URI.

   To emphasize what is already defined in [RFC3261], UASs MUST NOT use
   the "transport=tls" parameter.

5.2.  Registrar Behavior

   The UAC registers Contacts header fields to 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.

   A registrar MUST accept a binding to a SIPS Contact header field only
   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 and the Contacts
   and all the Path header fields, but does not include the From and To
   header fields.  If the URIs are not of the proper SIPS scheme, the
   registrar MUST reject the REGISTER with a 400 (Bad Request).

   A registrar can return a service route [RFC3608] and impose some
   constraints on whether or not TLS will be mandatory on specific hops.
   For example, if the topmost entry in the Path header field returned
   by the registrar is a SIPS URI, the registrar is telling the UAC that
   TLS is to be used for the first hop, even if the Request-URI is SIP.

   If a UA registered with a SIPS Contact header field, the registrar
   returning a service route [RFC3608] 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.

5.2.1.  GRUU

   When a GRUU [RFC5627] is assigned to an instance ID/AOR pair through
   registration, the registrar MUST assign both a SIP GRUU and a SIPS
   GRUU.  If the Contact header field in the REGISTER request contains a
   SIP URI, the registrar MUST return the SIP version of the GRUU.  If
   the Contact header field in the REGISTER request contains a SIPS URI,
   the registrar MUST return the SIPS version of the GRUU.

5.3.  Proxy Behavior

   Proxies MUST NOT use the last-hop exception of [RFC3261] 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 only 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 Contact header field.
   If the proxy needs to reject the request for that reason, the proxy
   MUST reject it with a 480 (Temporarily Unavailable) response.  In
   this case, the proxy SHOULD include a Warning header with warn-code
   380 "SIPS Not Allowed".

   Proxies SHOULD transport requests using a SIP URI over TLS when it is
   possible to set up a TLS connection, or reuse an existing one.
   [RFC5626], for example, allows for re-using an existing TLS
   connection.  Some proxies could have policies that prohibit 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, the proxy MUST replace that
   SIPS scheme with a SIP scheme while leaving the rest of the URI as
   is, and use the resulting URI as the Request-URI of the forwarded
   request.  The proxy MUST use TLS to forward the request to the UAS.
   Some proxies could have a policy of not forwarding at all requests
   using a non-SIPS Request-URI if the UAS had registered using a SIPS
   Contact header field.  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, the proxy MAY reject it with a 480
   (Temporarily Unavailable) response with a Warning header with warn-
   code 381 "SIPS Required".

   If a proxy needs to reject a request because the URIs are used
   inconsistently (e.g., the Request-URI is a SIPS URI, but the Contact
   header field is a SIP URI), the proxy SHOULD use response code 400
   (Bad Request).

   It is RECOMMENDED that the proxy use the outbound proxy procedures
   defined in [RFC5626] for supporting UACs that cannot 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, or a 416 response, or a
   480 (Temporarily Unavailable) response with a Warning header with
   warn-code 380 "SIPS Not Allowed" response, the proxy MUST NOT recurse
   on the response.  In this case, the proxy SHOULD forward the best
   response instead of recursing, in order to allow for the UAC to take
   the appropriate action.

   When a proxy sends a request using a SIP Request-URI and receives a
   3XX response with a SIPS Contact header field, or a 480 (Temporarily
   Unavailable) response with a Warning header with warn-code 381 "SIPS
   Required", the proxy MUST NOT recurse on the response.  In this case,
   the proxy SHOULD forward the best response instead of recursing, in
   order to allow for the UAC to take the appropriate action.

   To emphasize what is already defined in [RFC3261], proxies MUST NOT
   use the "transport=tls" parameter.

5.4.  Redirect Server Behavior

   Using a redirect server with TLS instead of using a proxy has some
   limitations that have to be taken into account.  Since there is no
   pre-established connection between the proxy and the UAS (such as
   with [RFC5626]), 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 TLS
   mutual authentication is used, and where there are no NAT traversal
   issues.  A redirect server would not be able to redirect to an entity
   that does not have a certificate.  A redirect server might not be
   usable 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 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 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 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 if it is in an environment
   where TLS is usable.  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, the UAS MAY reject the request with a 480 (Temporarily
   Unavailable) response with a Warning header with warn-code 380 "SIPS
   Not Allowed".  If a redirect server redirects to a UAS that it has no
   knowledge of (e.g., an AOR in a different domain), the Contact header
   field could be of any scheme.

   If a redirect server needs to reject a request because the URIs are
   used inconsistently (e.g., the Request-URI is a SIPS URI, but the
   Contact header field is a SIP URI), the redirect server SHOULD use
   response code 400 (Bad Request).

   To emphasize what is already defined in [RFC3261], redirect servers
   MUST NOT use the "transport=tls" parameter.

6.  Call Flows

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

                         example.com       .      example.net
                                           .
                       |-------------|     .    |------------|
                       | Registrar/  |__________|  Proxy  A  |
                       | Auth. Proxy |     .    |  (proxya)  |
                       |    (pb)     |     .    |------------|
                       |-------------|     .          |
                             |             .          |
                             |             .          |
                       |-----------|       .          |
                       |   Edge    |       .          |
                       |  Proxy B  |       .          |
                       |   (eb)    |       .          |
                       |-----------|       .          |
                        /        |         .          |
                       /         |         .          |
                      /          |         .          |
               ______            |         .          |
              |      |         _____       .        _____
              |______|        O / \ O      .       O / \ O
             /_______/         /___\       .        /___\
                                           .
             bob@bobpc      bob@bobphone   .         alice

                                 Topology

   In the following examples, Bob has two clients; one is a SIP PC
   client running on his computer, and the other one is a SIP phone.
   The PC client does not support SIPS, and 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 Edge Proxy B, and
   consequently, they include a Route header field indicating

   eb.example.com.  Edge Proxy B removes the Route header field
   corresponding to itself, and adds itself in a Path header field.  The
   registration process call flow is illustrated in Section 6.1.

   After registration, there are two 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 Proxy A.  Proxy A locates Bob's
   domain example.com.  In this example, that domain is owned by Bob's
   Registrar/Authoritative Proxy B.  Proxy A removes the Route header
   field corresponding to itself, and inserts itself in the Record-Route
   and forwards the request to Registrar/Authoritative Proxy B.

   The following subsections illustrate registration and three examples.
   In the first example (Section 6.2), Alice calls Bob's SIPS AOR.  In
   the second example (Section 6.3), Alice calls Bob's SIP AOR using TCP
   transport.  In the third example (Section 6.4), Alice calls Bob's SIP
   AOR using TLS transport.

6.1.  Bob Registers His Contacts

   This 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.

                                    (eb)           (pb)
                                    Edge        Registrar/
                Bob@bobpc          Proxy B     Auth. Proxy B
                 |                   |               |
                 |    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 -> Edge Proxy B

   REGISTER sip:pb.example.com SIP/2.0
   Via: SIP/2.0/TCP bobpc.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, outbound
   Route: <sip:eb.example.com;lr>
   Contact: <sip:bob@bobpc.example.com>
      ;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
      ;reg-id=1
   Content-Length: 0

   F2 REGISTER Edge Proxy B -> Registrar/Authoritative Proxy B

   REGISTER sip:pb.example.com SIP/2.0
   Via: SIP/2.0/TCP eb.example.com:5060;branch=z9hG4bK87asdks7
   Via: SIP/2.0/TCP bobpc.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, outbound
   Path: <sip:laksdyjanseg237+fsdf+uy623hytIJ8@eb.example.com;lr;ob>
   Contact: <sip:bob@bobpc.example.com>
      ;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
      ;reg-id=1
   Content-Length: 0

   F3 200 (REGISTER) Registrar/Authoritative Proxy B -> Edge Proxy B

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP eb.example.com:5060;branch=z9hG4bK87asdks7
   Via: SIP/2.0/TCP bobpc.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
   Require: outbound
   Supported: path, outbound
   Path: <sip:laksdyjanseg237+fsdf+uy623hytIJ8@eb.example.com;lr;ob>
   Contact: <sip:bob@bobphone.example.com>
      ;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
      ;reg-id=1
      ;expires=3600
   Date: Mon, 12 Jun 2006 16:43:12 GMT
   Content-Length: 0

   F4 200 (REGISTER) Edge Proxy B -> Bob's PC Client

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP bobpc.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
   Require: outbound
   Supported: path, outbound
   Path: <sip:laksdyjanseg237+fsdf+uy623hytIJ8@eb.example.com;lr;ob>
   Contact: <sip:bob@bobphone.example.com>
      ;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
      ;reg-id=1
      ;expires=3600
   Date: Thu, 09 Aug 2007 16:43:12 GMT
   Content-Length: 0

   F5 REGISTER Bob's Phone -> Edge Proxy B

   REGISTER sips:pb.example.com SIP/2.0
   Via: SIP/2.0/TLS bobphone.example.com:5061;branch=z9hG4bK9555
   Max-Forwards: 70
   To: Bob <sips:bob@example.com>
   From: Bob <sips:bob@example.com>;tag=90210
   Call-ID: faif9a@qwefnwdclk
   CSeq: 12 REGISTER
   Supported: path, outbound
   Route: <sips:eb.example.com;lr>
   Contact: <sips:bob@bobphone.example.com>
      ;+sip.instance="<urn:uuid:6F85D4E3-E8AA-46AA-B768-BF39D5912143>"
      ;reg-id=1
   Content-Length: 0

   F6 REGISTER Edge Proxy B -> Registrar/Authoritative Proxy B

   REGISTER sips:pb.example.com SIP/2.0
   Via: SIP/2.0/TLS eb.example.com:5061;branch=z9hG4bK876354
   Via: SIP/2.0/TLS bobphone.example.com:5061;branch=z9hG4bK9555
   Max-Forwards: 69
   To: Bob <sips:bob@example.com>
   From: Bob <sips:bob@example.com>;tag=90210
   Call-ID: faif9a@qwefnwdclk
   CSeq: 12 REGISTER
   Supported: path, outbound
   Path: <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>
   Contact: <sips:bob@bobphone.example.com>
      ;+sip.instance="<urn:uuid:6F85D4E3-E8AA-46AA-B768-BF39D5912143>"
      ;reg-id=1
   Content-Length: 0

   F7 200 (REGISTER) Registrar/Authoritative Proxy B -> Edge Proxy B

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS eb.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
   Require: outbound
   Supported: path, outbound
   Path: <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>
   Contact: <sips:bob@bobphone.example.com>
      ;+sip.instance="<urn:uuid:6F85D4E3-E8AA-46AA-B768-BF39D5912143>"
      ;reg-id=1
      ;expires=3600
   Date: Thu, 09 Aug 2007 16:43:50 GMT
   Content-Length: 0

   F8 200 (REGISTER) Edge 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
   Require: outbound
   Supported: path, outbound
   Path: <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>
   Contact: <sips:bob@bobphone.example.com>
      ;+sip.instance="<urn:uuid:6F85D4E3-E8AA-46AA-B768-BF39D5912143>"
      ;reg-id=1
      ;expires=3600
   Date: Thu, 09 Aug 2007 16:43:50 GMT
   Content-Length: 0

6.2.  Alice Calls Bob's SIPS AOR

   Bob's registration has already occurred as per Section 6.1.

   In this first example, Alice calls Bob's SIPS AOR
   (sips:bob@example.com).  Registrar/Authoritative Proxy B consults the
   binding in the registration database, and finds the two Contact
   header field bindings.  Alice had addressed Bob with a SIPS Request-
   URI (sips:bob@example.com), so Registrar/Authoritative Proxy B
   determines that the call needs to be routed only to bobphone (which
   registered using a SIPS Contact header field), and therefore the
   request is only sent to sips:bob@bobphone.example.com, through Edge
   Proxy B.  Both Registrar/Authoritative Proxy B and Edge Proxy B
   insert themselves in the Record-Route.  Bob answers at
   sips:bob@bobphone.example.com.

                           (eb)         (pb)
                           Edge      Registrar/
       Bob@bobpc          Proxy B   Auth. Proxy B   Proxy A     Alice
        |                   |            |            |            |
        |                   |            |            | INVITE F9  |
        |   Bob@bobphone    |            | INVITE F11 |<-----------|
        |      |            | INVITE F13 |<-----------|   100 F10  |
        |      | INVITE F15 |<-----------|   100 F12  |----------->|
        |      |<-----------|   100 F14  |----------->|            |
        |      |   180 F16  |----------->|            |            |
        |      |----------->|   180 F17  |            |            |
        |      |   200 F20  |----------->|   180 F18  |            |
        |      |----------->|   200 F21  |----------->|   180 F19  |
        |      |            |----------->|   200 F22  |----------->|
        |      |            |            |----------->|   200 F23  |
        |      |            |            |            |----------->|
        |      |            |            |            |   ACK F24  |
        |      |            |            |   ACK F25  |<-----------|
        |      |            |   ACK F26  |<-----------|            |
        |      |   ACK F27  |<-----------|            |            |
        |      |<-----------|            |            |            |
        |      |            |            |            |            |

                        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
   Route: <sips:proxya.example.net;lr>
   Contact: <sips:alice@alice-1.example.net>
   Content-Type: application/sdp
   Content-Length: {as per SDP}
   {SDP not shown}

   F10 100 (INVITE) Proxy A -> Alice

   SIP/2.0 100 Trying
   Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
   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 -> Registrar/Authoritative Proxy B

   INVITE sips:bob@example.com SIP/2.0
   Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
   Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
   Max-Forwards: 69
   To: Bob <sips:bob@example.com>
   From: Alice <sips:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route: <sips: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) Registrar/Authoritative Proxy B -> Proxy A

   SIP/2.0 100 Trying
   Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
   Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
   To: Bob <sips:bob@example.com>
   From: Alice <sips:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Content-Length: 0

   F13 INVITE Registrar/Authoritative Proxy B -> Edge Proxy B

   INVITE sips:bob@bobphone.example.com SIP/2.0
   Via: SIP/2.0/TLS pb.example.com:5061;branch=z9hG4bKbalouba
   Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
   Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
   Max-Forwards: 68
   To: Bob <sips:bob@example.com>
   From: Alice <sips:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Route:
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@edge.example.com;lr;ob>
   Record-Route: <sips:pb.example.com;lr>, <sips: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) Edge Proxy B -> Registrar/Authoritative Proxy B

   SIP/2.0 100 Trying
   Via: SIP/2.0/TLS pb.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

   F15 INVITE Edge Proxy B -> Bob's phone

   INVITE sips:bob@bobphone.example.com SIP/2.0
   Via: SIP/2.0/TLS eb.example.com:5061;branch=z9hG4bKbiba
   Via: SIP/2.0/TLS pb.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: 67
   To: Bob <sips:bob@example.com>
   From: Alice <sips:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
   Contact: <sips:alice@alice-1.example.net>
   Content-Type: application/sdp
   Content-Length: {as per SDP}
   {SDP not shown}

   F16 180 (INVITE) Bob's Phone -> Edge Proxy B

   SIP/2.0 180 Ringing
   Via: SIP/2.0/TLS eb.example.com:5061;branch=z9hG4bKbiba
   Via: SIP/2.0/TLS pb.example.com:5061;branch=z9hG4bKbalouba
   Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
   Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
   To: Bob <sips:bob@example.com>;tag=5551212
   From: Alice <sips:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
   Contact: <sips:bob@bobphone.example.com>
   Content-Length: 0

   F17 180 (INVITE) Edge Proxy B -> Registrar/Authoritative Proxy B

   SIP/2.0 180 Ringing
   Via: SIP/2.0/TLS pb.example.com:5061;branch=z9hG4bKbalouba
   Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
   Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
   To: Bob <sips:bob@example.com>;tag=5551212
   From: Alice <sips:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
   Contact: <sips:bob@bobphone.example.com>
   Content-Length: 0

   F18 180 Registrar/Authoritative 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 <sips:bob@example.com>;tag=5551212
   From: Alice <sips:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
   Contact: <sips:bob@bobphone.example.com>
   Content-Length: 0

   F19 180 (INVITE) Proxy A -> Alice

   SIP/2.0 180 Ringing
   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:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
   Contact: <sips:bob@bobphone.example.com>
   Content-Length: 0

   F20 200 (INVITE) Bob's Phone -> Edge Proxy B

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS eb.example.com:5061;branch=z9hG4bKbiba
   Via: SIP/2.0/TLS pb.example.com:5061;branch=z9hG4bKbalouba
   Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
   Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
   To: Bob <sips:bob@example.com>;tag=5551212
   From: Alice <sips:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
   Contact: <sips:bob@bobphone.example.com>
   Content-Length: 0

   F21 200 (INVITE) Edge Proxy B -> Registrar/Authoritative Proxy B

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS pb.example.com:5061;branch=z9hG4bKbalouba
   Via: SIP/2.0/TLS proxya.example.net:5061;branch=z9hG4bKpouet
   Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
   To: Bob <sips:bob@example.com>;tag=5551212
   From: Alice <sips:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
   Contact: <sips:bob@bobphone.example.com>
   Content-Length: 0

   F22 200 Registrar/Authoritative 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:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
   Contact: <sips:bob@bobphone.example.com>
   Content-Length: 0

   F23 200 (INVITE) Proxy A -> Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
   To: Bob <sips:bob@example.com>;tag=5551212
   From: Alice <sips:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
   Contact: <sips:bob@bobphone.example.com>
   Content-Length: 0

   F24 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:proxya.example.net;lr>, <sips:pb.example.com;lr>,
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@pb.example.com;lr;ob>
   Content-Length: 0

   F25 ACK Proxy A -> Registrar/Authoritative 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:pb.example.com;lr>,
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@pb.example.com;lr;ob>
   Content-Length: 0

   F26 ACK Registrar/Authoritative Proxy B -> Edge Proxy B

   ACK sips:bob@bobphone.example.com SIP/2.0
   Via: SIP/2.0/TLS pb.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: 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:pb.example.com;lr>,
    <sips:psodkfsj+34+kklsL+uJH-Xm816k09Kk@pb.example.com;lr;ob>
   Content-Length: 0

   F27 ACK Proxy B -> Bob's Phone

   ACK sips:bob@bobphone.example.com SIP/2.0
   Via: SIP/2.0/TLS eb.example.com:5061;branch=z9hG4bKkmfdgk
   Via: SIP/2.0/TLS pb.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

6.3.  Alice Calls Bob's SIP AOR Using TCP

   Bob's registration has already occurred as per Section 6.1.

   In the second example, Alice calls Bob's SIP AOR instead
   (sip:bob@example.com), and she uses TCP as a transport.  Registrar/
   Authoritative Proxy B consults the binding in the registration
   database, and finds the two Contact header field bindings.  Alice had
   addressed Bob with a SIP Request-URI (sip:bob@example.com), so
   Registrar/Authoritative Proxy B determines that the call needs to be
   routed both to bobpc (which registered with a SIP Contact header
   field) and bobphone (which registered with a SIPS Contact header
   field), and therefore the request is forked to
   sip:bob@bobpc.example.com and sip:bob@bobphone.example.com, through
   Edge Proxy B.  Note that Registrar/Authoritative Proxy B preserved
   the SIP scheme of the Request-URI instead of replacing it with the
   SIPS scheme of the Contact header field that was used for
   registration.  Both Registrar/Authoritative Proxy B and Edge Proxy B
   insert themselves in the Record-Route.  Bob's phone's policy is to
   accept calls to SIP and SIPS (i.e., "best effort"), so both his PC
   client and his SIP phone ring simultaneously.  Bob answers on his SIP
   phone, and the forked call leg to the PC client is canceled.

                           (eb)         (pb)
                           Edge      Registrar/
       Bob@bobpc          Proxy B   Auth. Proxy B   Proxy A     Alice
        |                   |            |            |            |
        |                   |            |            | INVITE F9  |
        |                   |            | INVITE F11 |<-----------|
        |                   | INVITE F13'|<-----------|   100 F10  |
        |    INVITE F15'    |<-----------|   100 F12  |----------->|
        |<------------------|   100 F14' |----------->|            |
        |     180 F16'      |----------->|            |            |
        |------------------>|   180 F17' |            |            |
        |                   |----------->|  180 F18'  |            |
        |   Bob@bobphone    |            |----------->|   180 F19' |
        |      |            | INVITE F13 |            |----------->|
        |      | INVITE F15 |<-----------|            |            |
        |      |<-----------|   100 F14  |            |            |
        |      |   180 F16  |----------->|            |            |
        |      |----------->|   180 F17  |            |            |
        |      |   200 F20  |----------->|   180 F18  |            |
        |      |----------->|   200 F21  |----------->|   180 F19  |
        |      |            |----------->|   200 F22  |----------->|
        |      |            |            |----------->|   200 F23  |
        |      |            |            |            |----------->|
        |      |            |            |            |   ACK F24  |
        |      |            |            |   ACK F25  |<-----------|
        |      |            |   ACK F26  |<-----------|            |
        |      |   ACK F27  |<-----------|            |            |
        |      |<-----------|            |            |            |
        |                   | CANCEL F26'|            |            |
        |    CANCEL F27'    |<-----------|            |            |
        |<------------------|            |            |            |
        |     200 F28'      |            |            |            |
        |------------------>|   200 F29' |            |            |
        |     487 F30'      |----------->|            |            |
        |------------------>|   487 F31' |            |            |
        |                   |----------->|            |            |

                         Alice Calls Bob's SIP AOR

   Message details

   F9 INVITE Alice -> Proxy A

   INVITE sip:bob@example.com SIP/2.0
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   Max-Forwards: 70
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Route: <sip:proxya.example.net;lr>
   Contact: <sip:alice@alice-1.example.net>
   Content-Type: application/sdp
   Content-Length: {as per SDP}
   {SDP not shown}

   F10 100 (INVITE) Proxy A -> Alice

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   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 -> Registrar/Authoritative Proxy B

   INVITE sip:bob@example.com SIP/2.0
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   Max-Forwards: 69
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route: <sip:proxya.example.net;lr>
   Contact: <sip:alice@alice-1.example.net>
   Content-Type: application/sdp
   Content-Length: {as per SDP}
   {SDP not shown}

   F12 100 (INVITE) Registrar/Authoritative 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 Registrar/Authoritative Proxy B -> Edge Proxy B

   INVITE sip:bob@bobpc.example.com SIP/2.0
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   Max-Forwards: 68
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Route: <sip:laksdyjanseg237+fsdf+uy623hytIJ8@eb.example.com;lr;ob>
   Record-Route: <sip:pb.example.com;lr>, <sip: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) Edge Proxy B -> Registrar/Authoritative Proxy B

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Content-Length: 0

   F15' INVITE Edge Proxy B -> Bob's PC Client

   INVITE sip:bob@bobpc.example.com SIP/2.0
   Via: SIP/2.0/TCP eb.example.com:5060;branch=z9hG4bKbiba
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   Max-Forwards: 67
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:laksdyjanseg237+fsdf+uy623hytIJ8@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:alice@alice-1.example.net>
   Content-Type: application/sdp
   Content-Length: {as per SDP}
   {SDP not shown}

   F16' 180 (INVITE) Bob's PC Client -> Edge Proxy B

   SIP/2.0 180 Ringing
   Via: SIP/2.0/TCP eb.example.com:5060;branch=z9hG4bKbiba
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>;tag=963258
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:laksdyjanseg237+fsdf+uy623hytIJ8@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobpc.example.com>
   Content-Length: 0

   F17' 180 (INVITE) Edge Proxy B -> Registrar/Authoritative Proxy B

   SIP/2.0 180 Ringing
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>;tag=963258
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:laksdyjanseg237+fsdf+uy623hytIJ8@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobpc.example.com>
   Content-Length: 0

   F18' 180 (INVITE) Registrar/Authoritative 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:laksdyjanseg237+fsdf+uy623hytIJ8@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobpc.example.com>
   Content-Length: 0

   F19' 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:laksdyjanseg237+fsdf+uy623hytIJ8@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobpc.example.com>
   Content-Length: 0

   F13 INVITE Registrar/Authoritative Proxy B -> Edge Proxy B

   INVITE sip:bob@bobphone.example.com SIP/2.0
   Via: SIP/2.0/TCP pb.example.com:5060;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
   Route: <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>
   Record-Route: <sip:pb.example.com;lr>, <sip: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) Edge Proxy B -> Registrar/Authoritative Proxy B

   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP pb.example.com:5060;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

   F15 INVITE Edge Proxy B -> Bob's Phone

   INVITE sip:bob@bobphone.example.com SIP/2.0
   Via: SIP/2.0/TLS eb.example.com:5061;branch=z9hG4bKtroubaba
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.1
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   Max-Forwards: 68
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:alice@alice-1.example.net>
   Content-Type: application/sdp
   Content-Length: {as per SDP}
   {SDP not shown}

   F16 180 (INVITE) Bob's Phone -> Edge Proxy B

   SIP/2.0 180 Ringing
   Via: SIP/2.0/TLS eb.example.com:5061;branch=z9hG4bKtroubaba
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.1
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>;tag=5551212
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobphone.example.com>
   Content-Length: 0

   F17 180 (INVITE) Edge Proxy B -> Registrar/Authoritative Proxy B

   SIP/2.0 180 Ringing
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.1
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>;tag=5551212
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobphone.example.com>
   Content-Length: 0

   F18 180 (INVITE) Registrar/Authoritative 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=5551212
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobphone.example.com>
   Content-Length: 0

   F19 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=5551212
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobphone.example.com>
   Content-Length: 0

   F20 200 (INVITE) Bob's Phone -> Edge Proxy B

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS eb.example.com:5061;branch=z9hG4bKtroubaba
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.1
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>;tag=5551212
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobphone.example.com>
   Content-Length: 0

   F21 200 (INVITE) Edge Proxy B -> Registrar/Authoritative Proxy B

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.1
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>;tag=5551212
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobphone.example.com>
   Content-Length: 0

   F22 200 (INVITE) Registrar/Authoritative Proxy B -> Proxy A

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>;tag=5551212
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobphone.example.com>
   Content-Length: 0

   F23 200 (INVITE) Proxy A -> Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>;tag=5551212
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Record-Route:
    <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>,
    <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
   Contact: <sip:bob@bobphone.example.com>
   Content-Length: 0

   F24 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:proxya.example.net;lr>, <sip:pb.example.com;lr>,
    <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@edge.example.com;lr;ob>
   Content-Length: 0

   F25 ACK Proxy A -> Registrar/Authoritative 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:pb.example.com;lr>,
          <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>
    Content-Length: 0

   F26 ACK Registrar/Authoritative Proxy B -> Edge Proxy B

   ACK sip:bob@bobphone.example.com SIP/2.0
   Via: SIP/2.0/TCP pb.example.com:5060;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: 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:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>
   Content-Length: 0

   F27 ACK Proxy B -> Bob's Phone

   ACK sip:bob@bobphone.example.com SIP/2.0
   Via: SIP/2.0/TLS eb.example.com:5061;branch=z9hG4bKtroubaba
   Via: SIP/2.0/TCP pb.example.com:5060;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

   F26' CANCEL Registrar/Authoritative Proxy B -> Edge Proxy B

   CANCEL sip:bob@bobpc.example.com SIP/2.0
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   Max-Forwards: 70
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 CANCEL
   Route: <sip:psodkfsj+34+kklsL+uJH-Xm816k09Kk@eb.example.com;lr;ob>
   Content-Length: 0

   F27' CANCEL Edge Proxy B -> Bob's PC Client

   CANCEL sip:bob@bobpc.example.com SIP/2.0
   Via: SIP/2.0/TCP eb.example.com:5060;branch=z9hG4bKtroubaba
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   Max-Forwards: 69
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 CANCEL
   Content-Length: 0

   F28' 200 (CANCEL) Bob's PC Client -> Edge Proxy B

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP eb.example.com:5060;branch=z9hG4bKtroubaba
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 CANCEL
   Content-Length: 0

   F29' 200 (CANCEL) Edge Proxy B -> Registrar/Authoritative Proxy B

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 CANCEL
   Content-Length: 0

   F30' 487 (INVITE) Bob's PC Client -> Edge Proxy B

   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/TCP eb.example.com:5060;branch=z9hG4bKtroubaba
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Content-Length: 0

   F31' 487 (INVITE) Edge Proxy B -> Registrar/Authoritative Proxy B

   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
   Via: SIP/2.0/TCP proxya.example.net:5060;branch=z9hG4bKpouet
   Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
   To: Bob <sip:bob@example.com>
   From: Alice <sip:alice@example.net>;tag=8675309
   Call-ID: lzksjf8723k@sodk6587
   CSeq: 1 INVITE
   Content-Length: 0

6.4.  Alice Calls Bob's SIP AOR Using TLS

   Bob's registration has already occurred as per Section 6.1.

   The third example is identical to the second one, except that Alice
   uses TLS as the transport for her connection to her proxy.  Such an
   arrangement would be common if Alice's UA supported TLS and wanted to
   use a single connection to the proxy (as would be the case when using
   [RFC5626]).  In the example below, Proxy A is also using TLS as a
   transport to communicate with Outbound Proxy B, but it is not
   necessarily the case.

   When using a SIP URI in the Request-URI but TLS as a transport for
   sending the request, the Via field indicates TLS.  The Route header
   field (if present) typically would use a SIP URI (but it could also
   be a SIPS URI).  The Contact header fields and To and From, however
   would also normally indicate a SIP URI.

   The call flow would be exactly as per the second example
   (Section 6.3).  The only difference would be that all the Via header
   fields would use TLS Via parameters.  The URIs would remain SIP URIs
   and not SIPS URIs.

7.  Further Considerations

   SIP [RFC3261] itself introduces some complications with using SIPS,
   for example, when Record-Route is not used.  When a SIPS URI is used
   in a Contact header field in a dialog-initiating request and Record-
   Route is not used, that SIPS URI might not be usable by the other
   end.  If the other end does not support SIPS and/or TLS, it will not
   be able to use it.  The last-hop exception is an example of when this
   can occur.  In this case, using Record-Route so that the requests are
   sent through proxies can help in making it work.  Another example is
   that even in a case where the Contact header field is a SIPS URI, no
   Record-Route is used, and the far end supports SIPS and TLS, it might
   still not be possible for the far end to establish a TLS connection
   with the SIP originating end if the certificate cannot be validated
   by the far end.  This could typically be the case if the originating
   end was using server-side authentication as described below, or if
   the originating end is not using a certificate that can be validated.

   TLS itself has a significant impact on how SIPS can be used.  Server-
   side authentication (where the server side provides its certificate
   but the client side does not) is typically used between a SIP end-
   user device acting as the TLS client side (e.g., a phone or a
   personal computer) and its SIP server (proxy or registrar) acting as
   the TLS server side.  TLS mutual authentication (where both the
   client side and the server side provide their respective
   certificates) is typically used between SIP servers (proxies,
   registrars), or statically configured devices such as PSTN gateways
   or media servers.  In the mutual authentication model, for two
   entities to be able to establish a TLS connection, it is required
   that both sides be able to validate each other's certificates, either
   by static configuration or by being able to recurse to a valid root
   certificate.  With server-side authentication, only the client side
   is capable of validating the server side's certificate, as the client
   side does not provide a certificate.  The consequences of all this
   are that whenever a SIPS URI is used to establish a TLS connection,
   it is expected to be possible for the entity establishing the
   connection (the client) to validate the certificate from the server
   side.  For server-side authentication, [RFC5626] is the recommended
   approach.  For mutual authentication, one needs to ensure that the
   architecture of the network is such that connections are made between
   entities that have access to each other's certificates.  Record-Route
   [RFC3261] and Path [RFC3327] are very useful in ensuring that
   previously established TLS connections can be reused.  Other
   mechanisms might also be used in certain circumstances: for example,
   using root certificates that are widely recognized allows for more
   easily created TLS connections.

8.  Security Considerations

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

   The "last-hop exception" of [RFC3261] introduced significant
   potential vulnerabilities in SIP, and it has therefore been
   deprecated by this specification.

   Section 26.4.4 of [RFC3261] describes the security considerations for
   the SIPS URI scheme.  These security considerations also applies
   here, as modified by Appendix A.

9.  IANA Considerations

   This specification registers two new warning codes, namely, 380 "SIPS
   Not Allowed" and 381 "SIPS Required".  The warning codes are defined
   as follows, and have been included in the Warning Codes (warn-codes)
   sub-registry of the SIP Parameters registry available from
   http://www.iana.org.

   380  SIPS Not Allowed: The UAS or proxy cannot process the request
        because the SIPS scheme is not allowed (e.g., because there are
        currently no registered SIPS contacts).

   381  SIPS Required: The UAS or proxy cannot process the request
        because the SIPS scheme is required.

   Reference: RFC 5630

   The note in the Warning Codes sub-registry is as follows:

      Warning codes provide information supplemental to the status code
      in SIP response messages.

10.  Acknowledgments

   The author would like to thank Jon Peterson, Cullen Jennings,
   Jonathan Rosenberg, John Elwell, Paul Kyzivat, Eric Rescorla, Robert
   Sparks, Rifaat Shekh-Yusef, Peter Reissner, Tina Tsou, Keith Drage,
   Brian Stucker, Patrick Ma, Lavis Zhou, Joel Halpern, Hisham
   Karthabil, Dean Willis, Eric Tremblay, Hans Persson, and Ben Campbell
   for their careful review and input.  Many thanks to Rohan Mahy for
   helping me with the subtleties of [RFC5626].

11.  References

11.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246, August 2008.

   [RFC5626]  Jennings, C., "Managing Client-Initiated Connections in
              the Session Initiation Protocol (SIP)", RFC 5626, October
              2009.

11.2.  Informative References

   [RFC2543]  Handley, M., Schulzrinne, H., Schooler, E., and J.
              Rosenberg, "SIP: Session Initiation Protocol", RFC 2543,
              March 1999.

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

   [RFC3515]  Sparks, R., "The Session Initiation Protocol (SIP) Refer
              Method", RFC 3515, April 2003.

   [RFC3608]  Willis, D. and B. Hoeneisen, "Session Initiation Protocol
              (SIP) Extension Header Field for Service Route Discovery
              During Registration", RFC 3608, October 2003.

   [RFC3725]  Rosenberg, J., Peterson, J., Schulzrinne, H., and G.
              Camarillo, "Best Current Practices for Third Party Call
              Control (3pcc) in the Session Initiation Protocol (SIP)",
              BCP 85, RFC 3725, April 2004.

   [RFC3891]  Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
              Protocol (SIP) "Replaces" Header", RFC 3891,
              September 2004.

   [RFC3893]  Peterson, J., "Session Initiation Protocol (SIP)
              Authenticated Identity Body (AIB) Format", RFC 3893,
              September 2004.

   [RFC3911]  Mahy, R. and D. Petrie, "The Session Initiation Protocol
              (SIP) "Join" Header", RFC 3911, October 2004.

   [RFC4168]  Rosenberg, J., Schulzrinne, H., and G. Camarillo, "The
              Stream Control Transmission Protocol (SCTP) as a Transport
              for the Session Initiation Protocol (SIP)", RFC 4168,
              October 2005.

   [RFC4244]  Barnes, M., "An Extension to the Session Initiation
              Protocol (SIP) for Request History Information", RFC 4244,
              November 2005.

   [RFC4474]  Peterson, J. and C. Jennings, "Enhancements for
              Authenticated Identity Management in the Session
              Initiation Protocol (SIP)", RFC 4474, August 2006.

   [RFC5627]  Rosenberg, J., "Obtaining and Using Globally Routable User
              Agent URIs (GRUU) in the Session Initiation Protocol
              (SIP)", RFC 5627, October 2009.

Appendix A.  Bug Fixes for RFC 3261

   In order to support the material in this document, this section makes
   corrections to RFC 3261.

   The last sentence of the fifth paragraph of Section 8.1.3.5 is
   replaced by:

      The client SHOULD retry the request, this time, using a SIP URI
      unless the original Request-URI used a SIPS scheme, in which case
      the client MUST NOT retry the request automatically.

   The fifth paragraph of Section 10.2.1 is replaced by:

      If the Address of Record in the To header field of a REGISTER
      request is a SIPS URI, then the UAC MUST also include only SIPS
      URIs in any Contact header field value in the requests.

   In Section 16.7 on p. 112 describing Record-Route, the second
   paragraph is deleted.

   The last paragraph of Section 19.1 is reworded as follows:

      A SIPS URI specifies that the resource be contacted securely.
      This means, in particular, that TLS is to be used on each hop
      between the UAC and the resource identified by the target SIPS
      URI.  Any resources described by a SIP URI (...)

   In the third paragraph of Section 20.43, the words "the session
   description" in the first sentence are replaced with "SIP".  Later in
   the paragraph, "390" is replaced with "380", and "miscellaneous
   warnings" is replaced with "miscellaneous SIP-related warnings".

   The second paragraph of Section 26.2.2 is reworded as follows:

      (...)  When used as the Request-URI of a request, the SIPS scheme
      signifies that each hop over which the request is forwarded, until
      the request reaches the resource identified by the Request-URI, is
      secured with TLS.  When used by the originator of a request (as
      would be the case if they employed a SIPS URI as the address-of-
      record of the target), SIPS dictates that the entire request path
      to the target domain be so secured.

   The first paragraph of Section 26.4.4 is replaced by the following:

      Actually using TLS on every segment of a request path entails that
      the terminating UAS is reachable over TLS (by registering with a
      SIPS URI as a contact address).  The SIPS scheme implies

      transitive trust.  Obviously, there is nothing that prevents
      proxies from cheating.  Thus, SIPS cannot guarantee that TLS usage
      will be truly respected end-to-end on each segment of a request
      path.  Note that since many UAs will not accept incoming TLS
      connections, even those UAs that do support TLS will be required
      to maintain persistent TLS connections as described in the TLS
      limitations section above in order to receive requests over TLS as
      a UAS.

   The first sentence of the third paragraph of Section 26.4.4 is
   replaced by the following:

      Ensuring that TLS will be used for all of the request segments up
      to the target UAS is somewhat complex.

   The fourth paragraph of Section 26.4.4 is deleted.

   The last sentence of the fifth paragraph of Section 26.4.4 is
   reworded as follows:

      S/MIME or, preferably, [RFC4474] may also be used by the
      originating UAC to help ensure that the original form of the To
      header field is carried end-to-end.

   In the third paragraph of Section 27.2, the phrase "when the failure
   of the transaction results from a Session Description Protocol (SDP)
   (RFC 2327 [1]) problem" is deleted.

   In the fifth paragraph of Section 27.2, "390" is replaced with "380",
   and "miscellaneous warnings" is replaced with "miscellaneous SIP-
   related warnings".

Author's Address

   Francois Audet
   Skype Labs

   EMail: francois.audet@skypelabs.com

 

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