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
  (http://trustee.ietf.org/license-info) in effect on the date of
  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
  outside the IETF Standards Process, and derivative works of it may
  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.



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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











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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.






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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].




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     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:[email protected];transport=TCP" and
     "sips:[email protected];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.



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  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



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  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.















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       ..........................         ...........................
       .                        .         .                         .
       .              +-------+ .         . +-------+               .
       .              |       | .         . |       |               .
       .              | 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



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  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.
























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       ..........................         ...........................
       .                        .         .                         .
       .              +-------+ .         . +-------+               .
       .              |       | .         . |       |               .
       .              | 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



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  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:[email protected] and sips:[email protected]) 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



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  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:[email protected] and sips:[email protected] 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:[email protected]
  or sips:[email protected] 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 [email protected]) registers with a
  SIPS Contact header field (e.g., sips:[email protected]), the
  registrar and the location service then know that Bob is reachable at
  sips:[email protected] and at sip:[email protected].





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  If a request is sent to AOR sips:[email protected], Bob's proxy will
  route it to Bob at Request-URI sips:[email protected].  If a
  request is sent to AOR sip:[email protected], Bob's proxy will route it
  to Bob at Request-URI sip:[email protected].

  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:[email protected]), then a request to AOR
  sips:[email protected] 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.




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  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



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  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.





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  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).















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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.



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  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



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  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.




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  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.





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  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




Audet                       Standards Track                    [Page 21]

RFC 5630                          SIPS                      October 2009


  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 [email protected]: sips:[email protected] and
  sip:[email protected].

  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.


























Audet                       Standards Track                    [Page 22]

RFC 5630                          SIPS                      October 2009


                                   (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:[email protected]>
  From: Bob <sip:[email protected]>;tag=456248
  Call-ID: 843817637684230@998sdasdh09
  CSeq: 1826 REGISTER
  Supported: path, outbound
  Route: <sip:eb.example.com;lr>
  Contact: <sip:[email protected]>
     ;+sip.instance="<urn:uuid:0C67446E-F1A1-11D9-94D3-000A95A0E128>"
     ;reg-id=1
  Content-Length: 0










Audet                       Standards Track                    [Page 23]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>
  From: Bob <sip:[email protected]>;tag=456248
  Call-ID: 843817637684230@998sdasdh09
  CSeq: 1826 REGISTER
  Supported: path, outbound
  Path: <sip:[email protected];lr;ob>
  Contact: <sip:[email protected]>
     ;+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:[email protected]>;tag=2493K59K9
  From: Bob <sip:[email protected]>;tag=456248
  Call-ID: 843817637684230@998sdasdh09
  CSeq: 1826 REGISTER
  Require: outbound
  Supported: path, outbound
  Path: <sip:[email protected];lr;ob>
  Contact: <sip:[email protected]>
     ;+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















Audet                       Standards Track                    [Page 24]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=2493K59K9
  From: Bob <sip:[email protected]>;tag=456248
  Call-ID: 843817637684230@998sdasdh09
  CSeq: 1826 REGISTER
  Require: outbound
  Supported: path, outbound
  Path: <sip:[email protected];lr;ob>
  Contact: <sip:[email protected]>
     ;+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:[email protected]>
  From: Bob <sips:[email protected]>;tag=90210
  Call-ID: faif9a@qwefnwdclk
  CSeq: 12 REGISTER
  Supported: path, outbound
  Route: <sips:eb.example.com;lr>
  Contact: <sips:[email protected]>
     ;+sip.instance="<urn:uuid:6F85D4E3-E8AA-46AA-B768-BF39D5912143>"
     ;reg-id=1
  Content-Length: 0

















Audet                       Standards Track                    [Page 25]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>
  From: Bob <sips:[email protected]>;tag=90210
  Call-ID: faif9a@qwefnwdclk
  CSeq: 12 REGISTER
  Supported: path, outbound
  Path: <sips:[email protected];lr;ob>
  Contact: <sips:[email protected]>
     ;+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:[email protected]>;tag=5150
  From: Bob <sips:[email protected]>;tag=90210
  Call-ID: faif9a@qwefnwdclk
  CSeq: 12 REGISTER
  Require: outbound
  Supported: path, outbound
  Path: <sips:[email protected];lr;ob>
  Contact: <sips:[email protected]>
     ;+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















Audet                       Standards Track                    [Page 26]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=5150
  From: Bob <sips:[email protected]>;tag=90210
  Call-ID: faif9a@qwefnwdclk
  CSeq: 12 REGISTER
  Require: outbound
  Supported: path, outbound
  Path: <sips:[email protected];lr;ob>
  Contact: <sips:[email protected]>
     ;+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:[email protected]).  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:[email protected]), 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:[email protected], through Edge
  Proxy B.  Both Registrar/Authoritative Proxy B and Edge Proxy B
  insert themselves in the Record-Route.  Bob answers at
  sips:[email protected].


















Audet                       Standards Track                    [Page 27]

RFC 5630                          SIPS                      October 2009


                          (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:[email protected] SIP/2.0
  Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKprout
  Max-Forwards: 70
  To: Bob <sips:[email protected]>
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Route: <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  Content-Type: application/sdp
  Content-Length: {as per SDP}
  {SDP not shown}










Audet                       Standards Track                    [Page 28]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Content-Length: 0


  F11 INVITE Proxy A -> Registrar/Authoritative Proxy B

  INVITE sips:[email protected] 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:[email protected]>
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route: <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  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:[email protected]>
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Content-Length: 0













Audet                       Standards Track                    [Page 29]

RFC 5630                          SIPS                      October 2009


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

  INVITE sips:[email protected] 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:[email protected]>
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Route:
   <sips:[email protected];lr;ob>
  Record-Route: <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  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:[email protected]>
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Content-Length: 0




















Audet                       Standards Track                    [Page 30]

RFC 5630                          SIPS                      October 2009


  F15 INVITE Edge Proxy B -> Bob's phone

  INVITE sips:[email protected] 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:[email protected]>
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sips:[email protected];lr;ob>,
   <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sips:[email protected];lr;ob>,
   <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  Content-Length: 0














Audet                       Standards Track                    [Page 31]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sips:[email protected];lr;ob>,
   <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sips:[email protected];lr;ob>,
   <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sips:[email protected];lr;ob>,
   <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  Content-Length: 0





Audet                       Standards Track                    [Page 32]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sips:[email protected];lr;ob>,
   <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sips:[email protected];lr;ob>,
   <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  Content-Length: 0


















Audet                       Standards Track                    [Page 33]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sips:[email protected];lr;ob>,
   <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sips:[email protected];lr;ob>,
   <sips:pb.example.com;lr>, <sips:proxya.example.net;lr>
  Contact: <sips:[email protected]>
  Content-Length: 0


  F24 ACK Alice -> Proxy A

  ACK sips:[email protected] SIP/2.0
  Via: SIP/2.0/TLS alice-1.example.net:5061;branch=z9hG4bKksdjf
  Max-Forwards: 70
  To: Bob <sips:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 ACK
  Route: <sips:proxya.example.net;lr>, <sips:pb.example.com;lr>,
   <sips:[email protected];lr;ob>
  Content-Length: 0








Audet                       Standards Track                    [Page 34]

RFC 5630                          SIPS                      October 2009


  F25 ACK Proxy A -> Registrar/Authoritative Proxy B

  ACK sips:[email protected] 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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 ACK
  Route: <sips:pb.example.com;lr>,
   <sips:[email protected];lr;ob>
  Content-Length: 0


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

  ACK sips:[email protected] 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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 ACK
  Route: <sips:pb.example.com;lr>,
   <sips:[email protected];lr;ob>
  Content-Length: 0


  F27 ACK Proxy B -> Bob's Phone

  ACK sips:[email protected] 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:[email protected]>;tag=5551212
  From: Alice <sips:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 ACK
  Content-Length: 0







Audet                       Standards Track                    [Page 35]

RFC 5630                          SIPS                      October 2009


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:[email protected]), 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:[email protected]), 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:[email protected] and sip:[email protected], 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.





























Audet                       Standards Track                    [Page 36]

RFC 5630                          SIPS                      October 2009


                          (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












Audet                       Standards Track                    [Page 37]

RFC 5630                          SIPS                      October 2009


  Message details

  F9 INVITE Alice -> Proxy A

  INVITE sip:[email protected] SIP/2.0
  Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
  Max-Forwards: 70
  To: Bob <sip:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Route: <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  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:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Content-Length: 0


  F11 INVITE Proxy A -> Registrar/Authoritative Proxy B

  INVITE sip:[email protected] 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:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route: <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  Content-Type: application/sdp
  Content-Length: {as per SDP}
  {SDP not shown}







Audet                       Standards Track                    [Page 38]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Content-Length: 0


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

  INVITE sip:[email protected] 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:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Route: <sip:[email protected];lr;ob>
  Record-Route: <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  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:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Content-Length: 0









Audet                       Standards Track                    [Page 39]

RFC 5630                          SIPS                      October 2009


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

  INVITE sip:[email protected] 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:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  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:[email protected]>;tag=963258
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  Content-Length: 0














Audet                       Standards Track                    [Page 40]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=963258
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  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:[email protected]>;tag=963258
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  Content-Length: 0




















Audet                       Standards Track                    [Page 41]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=963258
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  Content-Length: 0


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

  INVITE sip:[email protected] 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:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Route: <sip:[email protected];lr;ob>
  Record-Route: <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  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:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Content-Length: 0






Audet                       Standards Track                    [Page 42]

RFC 5630                          SIPS                      October 2009


  F15 INVITE Edge Proxy B -> Bob's Phone

  INVITE sip:[email protected] 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:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  Content-Length: 0














Audet                       Standards Track                    [Page 43]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  Content-Length: 0




















Audet                       Standards Track                    [Page 44]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  Content-Length: 0




















Audet                       Standards Track                    [Page 45]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  Content-Length: 0




















Audet                       Standards Track                    [Page 46]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 INVITE
  Record-Route:
   <sip:[email protected];lr;ob>,
   <sip:pb.example.com;lr>, <sip:proxya.example.net;lr>
  Contact: <sip:[email protected]>
  Content-Length: 0


  F24 ACK Alice -> Proxy A

  ACK sip:[email protected] SIP/2.0
  Via: SIP/2.0/TCP alice-1.example.net:5060;branch=z9hG4bKprout
  Max-Forwards: 70
  To: Bob <sip:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 ACK
  Route: <sip:proxya.example.net;lr>, <sip:pb.example.com;lr>,
   <sip:[email protected];lr;ob>
  Content-Length: 0


  F25 ACK Proxy A -> Registrar/Authoritative Proxy B

  ACK sip:[email protected] 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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 ACK
  Route: <sip:pb.example.com;lr>,
         <sip:[email protected];lr;ob>
   Content-Length: 0









Audet                       Standards Track                    [Page 47]

RFC 5630                          SIPS                      October 2009


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

  ACK sip:[email protected] 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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 ACK
  Route: <sip:[email protected];lr;ob>
  Content-Length: 0


  F27 ACK Proxy B -> Bob's Phone

  ACK sip:[email protected] 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:[email protected]>;tag=5551212
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 ACK
  Content-Length: 0


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

  CANCEL sip:[email protected] SIP/2.0
  Via: SIP/2.0/TCP pb.example.com:5060;branch=z9hG4bKbalouba.2
  Max-Forwards: 70
  To: Bob <sip:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 CANCEL
  Route: <sip:[email protected];lr;ob>
  Content-Length: 0










Audet                       Standards Track                    [Page 48]

RFC 5630                          SIPS                      October 2009


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

  CANCEL sip:[email protected] 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:[email protected]>
  From: Alice <sip:[email protected]>;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:[email protected]>
  From: Alice <sip:[email protected]>;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:[email protected]>
  From: Alice <sip:[email protected]>;tag=8675309
  Call-ID: lzksjf8723k@sodk6587
  CSeq: 1 CANCEL
  Content-Length: 0

















Audet                       Standards Track                    [Page 49]

RFC 5630                          SIPS                      October 2009


  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:[email protected]>
  From: Alice <sip:[email protected]>;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:[email protected]>
  From: Alice <sip:[email protected]>;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.



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RFC 5630                          SIPS                      October 2009


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.





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RFC 5630                          SIPS                      October 2009


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].







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RFC 5630                          SIPS                      October 2009


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.



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RFC 5630                          SIPS                      October 2009


  [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.
































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RFC 5630                          SIPS                      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



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RFC 5630                          SIPS                      October 2009


     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: [email protected]













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