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RFC 4458 - Session Initiation Protocol (SIP) URIs for Applicatio


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Network Working Group                                        C. Jennings
Request for Comments: 4458                                 Cisco Systems
Category: Informational                                         F. Audet
                                                         Nortel Networks
                                                               J. Elwell
                                                             Siemens plc
                                                              April 2006

        Session Initiation Protocol (SIP) URIs for Applications
         such as Voicemail and Interactive Voice Response (IVR)

Status of This Memo

   This memo provides information for the Internet community.  It does
   not specify an Internet standard of any kind.  Distribution of this
   memo is unlimited.

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
   redirecting targets from such applications.

Table of Contents

   1. Introduction ....................................................3
   2. Mechanism (User Agent Server 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 .....13
      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. Acknowledgements ...............................................18
   10. References ....................................................18
      10.1. Normative References .....................................18
      10.2. Informative References ...................................18

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
   voicemail 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, gateway, 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 Time Division Multiplexing
   (TDM)-based voicemail systems or to allow TDM systems to use VoIP
   unified messaging systems, this problem would be a little easier to
   solve.  The problem that is introduced in the Voice over IP (VoIP) to
   TDM case is as follows.  The SIP system needs to tell a Public
   Switched Telephone Network (PSTN) gateway 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 User Agent Clients
   (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.

2.  Mechanism (User Agent Server 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 that 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 voicemail system on the PSTN,
   the user portion will contain the phone number of the voicemail
   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 User Agent Server (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
   Private Branch Exchanges (PBXs) and voicemail systems, and for
   gateways that connect the IP network to the PSTN network.  Integrated
   Services Digital Network User Part (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
   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.

   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, UASes 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 on the behavior of downstream proxies; 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 in
   which the UM is not in the same administrative domain.

   This approach only works when the service that 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

   Note 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.

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 URIs 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 *

    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

    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 *

    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

    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 *

    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.

    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*

    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.

    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 Voice XML (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 an 800 number.
   The gateway 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 voicemail.

   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                 [RFC4458]
      cause           Yes                [RFC4458]

8.  Security Considerations

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

   The new URI parameters defined in this document 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 -- 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
   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 that there is 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 that 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 Hillary, when she tries to call Bill.  Several popular
   web portals will send SMS alert messages 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.

   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 form 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.  Acknowledgements

   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.

   [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.

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 plc
   Technology Drive
   Beeston, Nottingham  NG9 1LA
   UK

   EMail: john.elwell@siemens.com

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