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Versions: 00 01 02 03 04 05 06 RFC 4458

Network Working Group                                        C. Jennings
Internet-Draft                                             Cisco Systems
Expires: July 16, 2006                                          F. Audet
                                                         Nortel Networks
                                                               J. Elwell
                                                  Siemens Communications
                                                        January 12, 2006


    Session Initiation Protocol (SIP) URIs for Applications such as
             Voicemail and Interactive Voice Response (IVR)
                  draft-jennings-sip-voicemail-uri-06

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Copyright Notice

   Copyright (C) The Internet Society (2006).

Abstract

   The Session Initiation Protocol (SIP) is often used to initiate
   connections to applications such as voicemail or interactive voice
   recognition systems.  This specification describes a convention for
   forming SIP service URIs that request particular services based on



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   redirecting targets from such applications.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Mechanism (UAS and Proxy)  . . . . . . . . . . . . . . . . . .  4
     2.1.  Target . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.2.  Cause  . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.3.  Retrieving Messages  . . . . . . . . . . . . . . . . . . .  5
   3.  Interaction with Request History Information . . . . . . . . .  5
   4.  Limitations of Voicemail URI . . . . . . . . . . . . . . . . .  6
   5.  Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . . .  6
   6.  Examples . . . . . . . . . . . . . . . . . . . . . . . . . . .  7
     6.1.  Proxy Forwards Busy to Voicemail . . . . . . . . . . . . .  7
     6.2.  Endpoint forwards busy to Voicemail  . . . . . . . . . . .  9
     6.3.  Endpoint forwards busy to TDM via a gateway  . . . . . . . 11
     6.4.  Endpoint forwards busy to Voicemail with History Info  . . 12
     6.5.  Zero Configuration UM System . . . . . . . . . . . . . . . 14
     6.6.  Call Coverage  . . . . . . . . . . . . . . . . . . . . . . 15
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 15
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 16
     8.1.  Integrity Protection of Forwarding in SIP  . . . . . . . . 16
     8.2.  Privacy Related Issues on the Second Call Leg  . . . . . . 17
   9.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 18
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 18
     10.2. Informative References . . . . . . . . . . . . . . . . . . 18
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 20
   Intellectual Property and Copyright Statements . . . . . . . . . . 21





















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1.  Introduction

   Many applications such as Unified Messaging (UM systems and
   Interactive Voice Recognition (IVR) systems have been developed out
   of traditional telephony.  They can be used for storing and
   interacting with voice, video, faxes, email, and instant messaging
   services.  Users often use SIP to initiate communications with these
   applications.  When a SIP call is routed to an application, it is
   necessary that the application be able to obtain several bits of
   information from the session initiation message so that it can
   deliver the desired services.

   For the purpose of this document, we will use UM as the main example,
   but other applications may use the mechanism defined in this
   document.  The UM needs to know what mailbox should be used and
   possible reasons for the type of service desired from the UM.  Many
   voice mail systems provide different greetings depending whether the
   call went to voicemail because the user was busy or because the user
   did not answer.  All of this information can be delivered in existing
   SIP signaling from the call control that retargets the call to the
   UM, but there are no conventions for describing how the desired
   mailbox and the service requested are expressed.  It would be
   possible for every vendor to make this configurable so that any site
   could get it to work; however, this approach is unrealistic for
   achieving interoperability among call control, gateways, and unified
   messaging systems from different vendors.  This specification
   describes a convention for describing this mailbox and service
   information in the SIP URI so that vendors and operators can build
   interoperable systems.

   If there were no need to interoperate with TDM based voicemail
   systems or to allow TDM systems to use VoIP unified messaging
   systems, this problem would be a little easier.  The problem that is
   introduced in the VoIP to TDM case is as follows.  The SIP system
   needs to tell a PSTN GW both the subscriber's mailbox identifier
   (which typically looks like a phone number) and the address of the
   voicemail system in the TDM network (again a phone number).

   The question has been asked why the To header cannot be used to
   specify which mailbox to use.  One problem is that the call control
   proxies cannot modify the To header, and the UACs often set it
   incorrectly because they do not have information about the
   subscribers in the domain they are trying to call.  This happens
   because the routing of the call often translates the URI multiple
   times before it results in an identifier for the desired user that is
   valid in the namespace that the UM system understands.





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2.  Mechanism (UAS and Proxy)

   The mechanism works by encoding the information for the desired
   service in the SIP Request-URI that is sent to the UM system.  Two
   chunks of information are encoded, the first being the target mailbox
   to use and the second being the SIP status code that caused this
   retargeting and indicates the desired service.  The userinfo and
   hostport parts of the Request-URI will identify the voicemail
   service, the target mailbox can be put in the target parameter and
   the reason can be put in the cause parameter.  For example, if the
   proxy wished to use Bob's mailbox because his phone was busy, the URI
   sent to the UM system could be something like:

     sip:voicemail@example.com;target=bob%40example.com;cause=486

2.1.  Target

   Target is a URI parameter that indicates the address of the
   retargeting entity: in the context of UM, this can be the mailbox
   number.  For example, in the case of a voice mail system on the PSTN,
   the user portion will contain the phone number of the voice mail
   system, while the target will contain the phone number of the
   subscriber's mailbox.

2.2.  Cause

   Cause is a URI parameter that is used to indicate the service that
   the UAS receiving the message should perform.  The following values
   for this URI parameter are defined:

                +---------------------------------+-------+
                | Redirecting Reason              | Value |
                +---------------------------------+-------+
                | Unknown/Not available           | 404   |
                | User Busy                       | 486   |
                | No Reply                        | 408   |
                | Unconditional                   | 302   |
                | Deflection during alerting      | 487   |
                | Deflection immediate response   | 480   |
                | Mobile subscriber not reachable | 503   |
                +---------------------------------+-------+

   The mapping to PSTN protocols is important both for gateways that
   connect the IP network to existing TDM customer's equipment, such as
   PBXs and voicemail systems, and for gateways that connect the IP
   network to the PSTN network.  ISUP has signaling encodings for this
   information that can be treated as roughly equivalent for the
   purposes here.  For this reason, this specification uses the names of



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   Redirecting Reason values defined in ITU-T Q.732.2-5 [8].  In this
   specification, the Redirecting Reason Values are referred to as
   "Causes".  It should be understood that the term "Cause" has nothing
   to do with PSTN "Cause values" (as per ITU-T Q.850 [9] and RFC 3398
   [5]) but are instead mapped to ITU-T Q.732.2-5 Redirecting Reasons.
   Since ISUP interoperates with other PSTN networks such as Q.931 [10]
   and QSIG [11] using well-known rules, it makes sense to use the ISUP
   names as the most appropriate superset.  If no appropriate mapping to
   a cause value defined in this specification exists in a network, it
   would be mapped to 302 "Unconditional".  Similarly, if the mapping
   occurs from one of the causes defined in this specification to a PSTN
   system that does not have an equivalent reason value, it would be
   mapped to that network's equivalent of "Unconditional".  If a new
   cause parameter needs to be defined, this specification will have to
   be updated.

   The user portion of the URI SHOULD be used as the address of the
   voicemail system on the PSTN, while the target SHOULD be mapped to
   the original redirecting number on the PSTN side.

   The redirection counters SHOULD be set to one unless additional
   information is available.

2.3.  Retrieving Messages

   The UM system MAY use the fact that the From header is the same as
   the URI target as a hint that the user wishes to retrieve messages.


3.  Interaction with Request History Information

   The Request History mechanism [6] provides more information relating
   to multiple retargetings.  It is reasonable to have systems in which
   both the information in this specification and the History
   information are included and one or both are used.

   History-Info specifies a means of providing the UAS and UAC with
   information about the retargeting of a request.  This information
   includes the initial Request-URI and any retarget-to URIs.  This
   information is placed in the History-Info header field, which, except
   where prevented by privacy considerations, is built up as the request
   progresses and, upon reaching the UAS, is returned in certain
   responses.

   History-Info, when deployed at relevant SIP entities, is intended to
   provide a comprehensive trace of retargeting for a SIP request, along
   with the SIP response codes that led to retargeting.




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   History-Info can complement this specification.  In particular, when
   a proxy inserts a URI containing the parameters defined in this
   specification into the Request-URI of a forwarded request, the proxy
   can also insert a History-Info header field entry into the forwarded
   request and the URI in that entry will incorporate these parameters.
   Therefore even if the Request-URI is replaced as a result of
   rerouting by a downstream proxy, the History-Info header field will
   still contain these parameters, which may be of use to the UAS.
   Consequently, UAS that make use of this information may find the
   information in the History-Info header, and/or in the Request-URI,
   depending on the capability of the proxy to support generation of
   History-Info, or the behavior of downstream proxies, and therefore
   applications need to take this into account.


4.  Limitations of Voicemail URI

   This specification requires the proxy that is requesting the service
   to understand whether the UM system it is targeting supports the
   syntax defined in this specification.  Today, this information is
   provided to the proxy by configuration.  For practical purposes this
   means that the approach is unlikely to work in cases in which the
   proxy is not configured with information about the UM system or the
   UM is not in the same administrative domain.

   This approach only works when the service the call control wants
   applied is fairly simple.  For example it does not allow the proxy to
   express information like "Do not offer to connect to the target's
   colleague because that address has already been tried".

   The limitations discussed in this section are addressed by History-
   Info [6].


5.  Syntax

   The ABNF[4] grammar for these parameters is shown below.  The
   definitions of pvalue and Status-Code are defined in the ABNF in RFC
   3261[1].

     target-param      =  "target" EQUAL pvalue

     cause-param       =  "cause" EQUAL Status-Code

   It is worth noting that the ABNF requires some characters to be
   escaped if they occur in the value of the target parameters.  For
   example, the "@" character needs to be escaped.




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6.  Examples

   This section provides some example use cases for the solution
   proposed in this document.  For the purpose of this document, UM is
   used as the main example, but other applications may use this
   mechanism.  The examples are intended to highlight the potential
   applicability of this solution and are not intended to limit its
   applicability.

   Also the examples show just service retargeting on busy, but can
   easily be adapted to show other forms of retargeting.

   In several of the examples, the URI are broken across more than one
   line.  This was only done for formatting and is not a valid SIP
   message.  Some of the characters in the URIs are not correctly
   escaped to improve readability.  The examples are all shown using
   sip: with UDP transport, for readability.  It should be understood
   that using sips: with TLS transport is preferable.

6.1.  Proxy Forwards Busy to Voicemail

   In this example, Alice calls Bob. Bob's proxy determines that Bob is
   busy, and the proxy forwards the call to Bob's voicemail.  Alice's
   phone is at 192.0.2.1 while Bob's phone is at 192.0.2.2.  The
   important thing to note is the URI in message F7.

     Alice            Proxy           Bob             voicemail
       |                |              |                   |
       |    INVITE F1   |              |                   |
       |--------------->|   INVITE F2  |                   |
       |                |------------->|                   |
       |(100 Trying) F3 |              |                   |
       |<---------------|  486 Busy F4 |                   |
       |                |<-------------|                   |
       |                |     ACK F5   |                   |
       |                |------------->|                   |
       |(181 Call is Being Forwarded) F6                   |
       |<---------------|              |    INVITE F7      |
       |                |--------------------------------->|
                    * Rest of flow not shown *











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    F1: INVITE 192.0.2.1 -> proxy.example.com

    INVITE sip:+15555551002@example.com;user=phone  SIP/2.0
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:alice@192.0.2.1>
    Content-Type: application/sdp
    Content-Length: *Body length goes here*

    * SDP goes here*


    F2: INVITE proxy.example.com -> 192.0.2.2

    INVITE sip:+15555551002@192.0.2.2 SIP/2.0
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-1
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:alice@192.0.2.1>
    Content-Type: application/sdp
    Content-Length: *Body length goes here*

    * SDP goes here*


    F4: 486 192.0.2.2 -> proxy.example.com

    SIP/2.0 486 Busy Here
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-1
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone;tag=09xde23d80
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Content-Length: 0








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    F7: INVITE proxy.example.com -> um.example.com

    INVITE sip:voicemail@example.com;\
           target=sip:+15555551002%40example.com;user=phone;\
           cause=486  SIP/2.0
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-2
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:alice@192.0.2.1>
    Content-Type: application/sdp
    Content-Length: *Body length goes here*

    * SDP goes here*

6.2.  Endpoint forwards busy to Voicemail

   In this example, Alice calls Bob. Bob is busy, but forwards the
   session directly to his voicemail.  Alice's phone is at 192.0.2.1
   while Bob's phone is at 192.0.2.2.  The important thing to note is
   the URI in the Contact in message F3.

     Alice            Proxy           Bob             voicemail
       |                |              |                   |
       |    INVITE F1   |              |                   |
       |--------------->|   INVITE F2  |                   |
       |                |------------->|                   |
       |                | 302 Moved F3 |                   |
       |  302 Moved  F4 |<-------------|                   |
       |<---------------|              |                   |
       |      ACK F5    |              |                   |
       |--------------->|     ACK F6   |                   |
       |                |------------->|                   |
       |                      INVITE F7                    |
       |-------------------------------------------------->|
                   * Rest of flow not shown *












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    F1: INVITE 192.0.2.1 -> proxy.example.com

    INVITE sip:+15555551002@example.com;user=phone  SIP/2.0
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:alice@192.0.2.1>
    Content-Type: application/sdp
    Content-Length: *Body length goes here*

    * SDP goes here*


    F2: INVITE proxy.example.com -> 192.0.2.2

    INVITE sip:line1@192.0.2.2 SIP/2.0
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-1
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:alice@192.0.2.1>
    Content-Type: application/sdp
    Content-Length: *Body length goes here*

    * SDP goes here*


    F3: 302 192.0.2.2 -> proxy.example.com

    SIP/2.0 302 Moved Temporarily
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-1
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone;tag=09xde23d80
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Contact: <sip: voicemail@example.com;\
           target=sip:+15555551002%40example.com;user=phone;\
           cause=486;>
    Content-Length: 0





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    F7: INVITE proxy.example.com -> um.example.com

    INVITE sip: voicemail@example.com;\
           target=sip:+15555551002%40example.com;user=phone;\
           cause=486  SIP/2.0
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-2
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:alice@192.0.2.1>
    Content-Type: application/sdp
    Content-Length: *Body length goes here*

    * SDP goes here*

6.3.  Endpoint forwards busy to TDM via a gateway

   In this example, the voicemail is reached via a gateway to a TDM
   network.  Bob's number is +1 555 555-1002, while voicemail's number
   on the TDM network is +1-555-555-2000.

   The call flow is the same as in Section 6.2 except for the Contact
   URI in F4 and the Request URI in F7.

     Alice            Proxy           Bob             voicemail
       |                |              |                   |
       |    INVITE F1   |              |                   |
       |--------------->|   INVITE F2  |                   |
       |                |------------->|                   |
       |(100 Trying) F3 |              |                   |
       |<---------------| 302 Moved F4 |                   |
       |                |<-------------|                   |
       |                |     ACK F5   |                   |
       |                |------------->|                   |
       |(181 Call is Being Forwarded) F6                   |
       |<---------------|              |    INVITE F7      |
       |                |--------------------------------->|
                    * Rest of flow not shown *










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    F4: 486 192.0.2.2 -> proxy.example.com

    SIP/2.0 302 Moved temporarily
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-1
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone;tag=09xde23d80
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Contact: <sip:+15555552000@example.com;user=phone;\
              target=tel:+15555551002;cause=486>
    Content-Length: 0


    F7: INVITE proxy.example.com -> gw.example.com

    INVITE sip:+15555552000@example.com;user=phone;\
           target=tel:+15555551002;cause=486\
           SIP/2.0
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-2
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:alice@192.0.2.1;transport=tcp>
    Content-Type: application/sdp
    Content-Length: *Body length goes here*

    * SDP goes here*

6.4.  Endpoint forwards busy to Voicemail with History Info

   This example illustrates how History Info works in conjunction with
   service retargeting.  The scenario is the same as Section 6.1.















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       F1: INVITE 192.0.2.1 -> proxy.example.com

       INVITE sip:+15555551002@example.com;user=phone  SIP/2.0
       Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
       From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
       To: sip:+15555551002@example.com;user=phone
       Call-ID: c3x842276298220188511
       CSeq: 1 INVITE
       Max-Forwards: 70
       Contact: <sip:alice@192.0.2.1>
       History-Info: <sip:+15555551002@example.com;user=phone >;index=1
       Content-Type: application/sdp
       Content-Length: *Body length goes here*

       * SDP goes here*


    F2: INVITE proxy.example.com -> 192.0.2.2

    INVITE sip:line1@192.0.2.2 SIP/2.0
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-1
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:alice@192.0.2.1>
    History-Info: <sip:+15555551002@example.com;user=phone >;index=1,
                  <sip:line1@192.0.2.4>;index=1.1

    Content-Type: application/sdp
    Content-Length: *Body length goes here*

    * SDP goes here*
















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    F7: INVITE proxy.example.com -> um.example.com

    INVITE sip: voicemail@example.com;\
           target=sip:+15555551002%40example.com;user=phone;\
           cause=486  SIP/2.0
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-2
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:alice@192.0.2.1>
    History-Info: <sip:+15555551002@example.com;user=phone >;index=1,
                  <sip:line1@192.0.2.4?Reason=SIP%3Bcause%3D302;\
                   text="Moved Temporarily">;index=1.1
                  <sip: voicemail@example.com;\
                   target=sip:+15555551002%40example.com;user=phone;\
                   cause=486>;index=2
    Contact: <sip:alice@192.0.2.1>
    Content-Type: application/sdp
    Content-Length: *Body length goes here*

    * SDP goes here*

6.5.  Zero Configuration UM System

   In this example, the UM system has no configuration information
   specific to any user.  The proxy is configured to pass a URI that
   provides the prompt to play and an email address in the user portion
   of the URI to which the recorded message is to be sent.

   The call flow is the same as in Section 6.1, except that the URI in
   F7 changes to specify the user part as Bob's email address, and the
   Netann [7] URI play parameter specifies where the greeting to play
   can be fetched from.















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    F7: INVITE proxy.example.com -> voicemail.example.com

    INVITE sip:voicemail@example.com;target=mailto:bob%40example.com;\
       cause=486;play=http://www.example.com/bob/busy.wav SIP/2.0
    Via: SIP/2.0/TCP 192.0.2.4:5060;branch=z9hG4bK-ik80k7g-2
    Via: SIP/2.0/TCP 192.0.2.1:5060;branch=z9hG4bK-74bf9
    From: Alice <sip:+15555551001@example.com;user=phone>;tag=9fxced76sl
    To: sip:+15555551002@example.com;user=phone
    Call-ID: c3x842276298220188511
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:alice@192.0.2.1>
    Content-Type: application/sdp
    Content-Length: *Body length goes here*

    * SDP goes here*

   In addition, if the proxy wished to indicate a VXML script that the
   UM should execute, it could add a parameter to the URI in the above
   message that looked like:

    voicexml=http://www.example.com/bob/busy.vxml

6.6.  Call Coverage

   In a Call Coverage example, a user on the PSTN calls a 800 number.
   The GW sends this to the proxy which recognizes that the helpdesk is
   the target.  Alice and Bob are staffing the help desk and are tried
   sequentially but neither answers, so the call is forwarded to the
   helpdesk's voice mail.

   The details of this flow are trivial and not shown: the key item in
   this example is that the INVITE to Alice and Bob looks as follows:

     INVITE sip:voicemail@example.com;target=helpdesk%40example.com;\
            cause=302 SIP/2.0


7.  IANA Considerations

   This specification adds two new values to the IANA registration in
   the "SIP/SIPS URI Parameters" registry as defined in [3].

      Parameter Name  Predefined Values  Reference
      ____________________________________________
      target          No                 [RFCAAAA]
      cause           Yes                [RFCAAAA]




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   [Note to IANA: Please replace AAAA with the RFC number of this
   specification.


8.  Security Considerations

   This draft discusses transactions involving at least three parties,
   which increases the complexity of the privacy issues.

   The new URI parameters defined in this draft are generally sent from
   a Proxy or call control system to a Unified Messaging (UM) system or
   to a gateway to the PSTN and then to a voicemail system.  These new
   parameters tell the UM what service the proxy wishes to have
   performed.  Just as any message sent from the proxy to the UM needs
   to be integrity protected, these messages need to be integrity
   protected to stop attackers from, for example, causing a voicemail
   meant for a company's CEO to go to an attacker's mailbox.  RFC 3261
   provides a TLS mechanism suitable for performing this integrity
   protection.

   The signaling from the Proxy to the UM or gateway will reveal who is
   calling whom and possibly some information about a user's presence
   based on whether the call was answered or sent to voicemail.  This
   information can be protected by encrypting the SIP traffic between
   the Proxy and UM or gateway.  Again, RFC 3261 contains mechanisms for
   accomplishing this using TLS.

   Implementations should implement and use TLS.

8.1.  Integrity Protection of Forwarding in SIP

   The forwarding of a call in SIP brings up a very strange trust issue.
   Consider the normal case when A calls B and the call gets forwarded
   to C by a network element in B's domain, and then C answers the call.
   A has called B but ended up talking to C. This scenario may be hard
   to separate from a man-in-the-middle attack.

   There are two possible solutions.  One is that B sends back
   information to A saying don't call me, call C and signs it as B. The
   problem is that this solution involves revealing that B has forwarded
   to C, which B often may not want to do.  For example, B may be a work
   phone that has been forwarded to a mobile or home phone.  The user
   does not want to reveal their mobile or home phone number but, even
   more importantly, does not want to reveal that they are not in the
   office.

   The other possible solution is that A needs to trust B only to
   forward to a trusted identity.  This requires a hop-by-hop transitive



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   trust such that each hop will only send to a trusted next hop and
   each hop will only do things that the user at that hop desired.  This
   solution is enforced in SIP using the SIPS URI and TLS based hop-by-
   hop security.  It protects from an off axis attack, but if one of the
   hops is not trustworthy, the call may be diverted to an attacker.

   Any redirection of a call to an attacker's mailbox is serious.  It is
   trivial for an attacker to make its mailbox seem very much like the
   real mailbox and forward the messages to the real mailbox so that the
   fact that the messages have been intercepted or even tampered with
   escapes detection.  Approaches such as the SIPS URL and the History-
   Info[6] can help protect against these attacks.

8.2.  Privacy Related Issues on the Second Call Leg

   In the case where A calls B and gets redirected to C, occasionally
   people suggest there is a a requirement for the call leg from B to C
   to be anonymous.  The SIP case is not the PSTN and there is no call
   leg from B to C; instead there is a VoIP session between A and C. If
   A has put a To header field value containing B in the initial invite
   message, unless something special is done about it, C would see that
   To header field value.  If the person who answers phone C says "I
   think you dialed the wrong number, who were you trying to reach?"  A
   will probably specify B.

   If A does not want C to see that the call was to B, A needs a special
   relationship with the forwarding Proxy to induce it not to reveal
   that information.  The call should go through an anonymization
   service that provides session or user level privacy (as described in
   RFC 3323 [2]) service before going to C. It is not hard to figure out
   how to meet this requirement, but it is unclear why anyone would want
   this service.

   The scenario in which B wants to make sure that C does not see that
   the call was to B is easier to deal with but a bit weird.  The usual
   argument is Bill wants to forward his phone to Monica but does not
   want Monica to find out his phone number.  It is hard to imagine that
   Monica would want to accept all Bill's calls without knowing how to
   call Bill to complain.  The only person Monica will be able to
   complain to is Hilary, when she tries to call Bill.  Several popular
   web portals will send SMS alert message about things like stock
   prices and weather to mobile phone users today.  Some of these
   contain no information about the account on the web portal that
   initiated them, making it nearly impossible for the mobile phone
   owner to stop them.  This anonymous message forwarding has turned out
   to be a really bad idea even where no malice is present.  Clearly
   some people are fairly dubious about the need for this, but never
   mind: let's look at how it is solved.



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   In the general case, the proxy needs to route the call through an
   anonymization service and everything will be cleaned up.  Any
   anonymization service that performs the "Privacy: Header" Service in
   RFC 3323 [2] must remove the cause and target URI parameters from the
   URI.  Privacy of the parameters when they from part of a URI within
   the History-Info header is covered in History-Info [6].

   This specification does not discuss the security considerations of
   mapping to a PSTN Gateway.  Security implications of mapping to ISUP
   for example, are discussed in RFC 3398 [5].


9.  Acknowledgments

   Many thanks to Mary Barnes, Steve Levy, Dean Willis, Allison Mankin,
   Martin Dolly, Paul Kyzivat, Erick Sasaki, Lyndsay Campbell, Keith
   Drage, Miguel Garcia, Sebastien Garcin, Roland Jesske, Takumi Ohba
   and Rohan Mahy.


10.  References

10.1.  Normative References

   [1]  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.

   [2]  Peterson, J., "A Privacy Mechanism for the Session Initiation
        Protocol (SIP)", RFC 3323, November 2002.

   [3]  Camarillo, G., "The Internet Assigned Number Authority (IANA)
        Uniform Resource Identifier (URI) Parameter Registry for the
        Session Initiation Protocol (SIP)", BCP 99, RFC 3969,
        December 2004.

   [4]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
        Specifications: ABNF", RFC 4234, October 2005.

10.2.  Informative References

   [5]   Camarillo, G., Roach, A., Peterson, J., and L. Ong, "Integrated
         Services Digital Network (ISDN) User Part (ISUP) to Session
         Initiation Protocol (SIP) Mapping", RFC 3398, December 2002.

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



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   [7]   Burger, E., Van Dyke, J., and A. Spitzer, "Basic Network Media
         Services with SIP", RFC 4240, December 2005.

   [8]   "Stage 3 description for call offering supplementary services
         using signalling system No. 7: Call diversion services", ITU-
         T Recommendation Q.732.2-5, December 1999.

   [9]   "Usage of cause and location in the Digital Subscriber
         Signalling System No. 1 and the Signalling System No. 7 ISDN
         User Part", ITU-T Recommendation Q.850, May 1998.

   [10]  "ISDN user-network interface layer 3 specification for basic
         call control", ITU-T Recommendation Q.931, May 1998.

   [11]  "Information technology - Telecommunications and information
         exchange between systems - Private Integrated Services Network
         - Circuit mode bearer services - Inter-exchange signalling
         procedures and protocol", ISO/IEC 11572, March 2000.

































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Authors' Addresses

   Cullen Jennings
   Cisco Systems
   170 West Tasman Drive
   Mailstop SJC-21/2
   San Jose, CA  95134
   USA

   Phone: +1 408 421-9990
   Email: fluffy@cisco.com


   Francois Audet
   Nortel Networks
   4655 Great America Parkway
   Santa Clara, CA  95054
   US

   Phone: +1 408 495 3756
   Email: audet@nortel.com


   John Elwell
   Siemens Communications
   Technology Drive
   Beeston, Nottingham  NG9 1LA
   UK

   Email: john.elwell@siemens.com





















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