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Dispatch                                                       D. Worley
Internet-Draft                                                     Avaya
Intended status: Informational                         December 29, 2010
Expires: July 2, 2011


      Session Initiation Protocol Service Example -- Music on Hold
                    draft-worley-service-example-06

Abstract

   The "music on hold" feature is one of the most desired features of
   telephone systems in the business environment.  "Music on hold" is
   where, when one party to a call has the call "on hold", that party's
   telephone provides an audio stream (often music) to be heard by the
   other party.  Architectural features of SIP make it difficult to
   implement music-on-hold in a way that is fully compliant with the
   standards.  The implementation of music-on-hold described in this
   document is fully effective and standards-compliant, and has a number
   of advantages over the methods previously documented.  In particular,
   it is less likely to produce peculiar user interface effects and more
   likely to work in systems which perform authentication than the
   method of RFC 5359.

Status of this Memo

   This Internet-Draft is submitted to IETF in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on July 2, 2011.

Copyright Notice

   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents



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   (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 Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
     1.1.  Intended Status  . . . . . . . . . . . . . . . . . . . . .  4
   2.  Technique  . . . . . . . . . . . . . . . . . . . . . . . . . .  5
     2.1.  Placing a Call on Hold and Establishing an External
           Media Stream . . . . . . . . . . . . . . . . . . . . . . .  5
     2.2.  Taking a Call off Hold and Terminating the External
           Media Stream . . . . . . . . . . . . . . . . . . . . . . .  6
     2.3.  Example Message Flow . . . . . . . . . . . . . . . . . . .  7
     2.4.  Re-INVITE and UPDATE from the Remote UA  . . . . . . . . . 15
     2.5.  INVITE with Replaces . . . . . . . . . . . . . . . . . . . 16
     2.6.  Re-INVITE and UPDATE from the Music-On-Hold Source . . . . 16
     2.7.  Handling Payload Type Numbers  . . . . . . . . . . . . . . 17
       2.7.1.  Analysis . . . . . . . . . . . . . . . . . . . . . . . 17
       2.7.2.  Solution to the Problem  . . . . . . . . . . . . . . . 18
       2.7.3.  Example of the Solution  . . . . . . . . . . . . . . . 19
     2.8.  Dialog/Session Timers  . . . . . . . . . . . . . . . . . . 22
   3.  Advantages . . . . . . . . . . . . . . . . . . . . . . . . . . 23
   4.  Caveats  . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
     4.1.  Offering All Available Media Formats . . . . . . . . . . . 24
     4.2.  Handling re-INVITES in a B2BUA . . . . . . . . . . . . . . 24
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 26
   6.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 27
   7.  Revision History . . . . . . . . . . . . . . . . . . . . . . . 28
     7.1.  Changes from draft-worley-service-example-00 to
           draft-worley-service-example-01  . . . . . . . . . . . . . 28
     7.2.  Changes from draft-worley-service-example-01 to
           draft-worley-service-example-02  . . . . . . . . . . . . . 28
     7.3.  Changes from draft-worley-service-example-02 to
           draft-worley-service-example-03  . . . . . . . . . . . . . 28
     7.4.  Changes from draft-worley-service-example-03 to
           draft-worley-service-example-04  . . . . . . . . . . . . . 28
     7.5.  Changes from draft-worley-service-example-04 to
           draft-worley-service-example-05  . . . . . . . . . . . . . 29
     7.6.  Changes from draft-worley-service-example-05 to
           draft-worley-service-example-06  . . . . . . . . . . . . . 29
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 30
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 30



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     8.2.  Informative References . . . . . . . . . . . . . . . . . . 30
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 32

















































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1.  Introduction

   Within SIP[sip]-based systems, it is desirable to be able to provide
   features that are similar to those provided by traditional telephony
   systems.  A frequently requested feature is "music on hold": The
   music-on-hold feature is where, when one party to a call has the call
   "on hold", that party's telephone provides an audio stream (often
   music) to be heard by the other party.

   Architectural features of SIP make it difficult to implement music-
   on-hold in a way that is fully compliant with the standards.  The
   purpose of this document is to describe a method that is reasonably
   simple yet fully effective and standards-compliant.

1.1.  Intended Status

   The "intended status" of this document is "Informational".  The
   reason that it is not "Best Current Practice" is that this method is
   not specified as "best", nor is this specification intended to
   supersede all other methods for implementing music-on-hold.  Indeed,
   the two user agents in a call can use different methods for
   implementing music-on-hold, as can different user agents within a
   telephone system.




























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2.  Technique

   The essence of the technique is that when the executing UA (the
   user's UA) performs a re-INVITE of the remote UA (the other user's
   UA) to establish the hold state, it provides no SDP[sdp]
   offer[offer-answer][offer-answer-bis], thus compelling the remote UA
   to provide an SDP offer.  The executing UA then extracts the offer
   SDP from the remote UA's 2xx response, and uses that as the offer SDP
   in a new INVITE to the external media source.  The external media
   source is thus directed to provide media directly to the remote UA.
   The media source's answer SDP is returned to the remote UA in the ACK
   to the re-INVITE.

2.1.  Placing a Call on Hold and Establishing an External Media Stream

   1.  The executing user instructs the executing UA to put the dialog
       on-hold.

   2.  The executing UA sends a re-INVITE without SDP to the remote UA,
       which forces the remote UA to provide an SDP offer in its 2xx
       response.  The Contact header of the re-INVITE includes the
       '+sip.rendering="no"' field parameter to indicate that it is
       putting the call on hold. ([dialog-event] section 5.2)

   3.  The remote UA sends a 2xx to the re-INVITE, and includes an SDP
       offer giving its own listening address/port.  If the remote UA
       understands the sip.rendering feature parameter, the offer may
       indicate that it will not send media by specifying the media
       directionalities as "recvonly" (the reverse of "on-hold") or
       "inactive".  But the remote UA may offer to send media.

   4.  The executing UA uses this offer to derive the offer SDP of an
       initial INVITE that it sends to the configured music-on-hold
       (MOH) source.  The SDP in this request is largely copied from the
       SDP returned by the remote UA in the previous step, particularly
       regarding the provided listening address/port and payload type
       numbers.  But the media directionalities are restricted to
       "recvonly" or "inactive" as appropriate.  The executing UA may
       want or need to change the o= line.  In addition, some a=rtpmap
       lines may need to be added to control the assignment of RTP
       payload type numbers.[Section 2.7]

   5.  The MOH source sends a 2xx response to the INVITE, which contains
       an SDP answer that should include its media source address as its
       listening address/port.  This SDP must necessarily[offer-answer]
       specify "sendonly" or "inactive" as the directionality for all
       media streams.




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       Although this address/port should receive no RTP, the specified
       port determines the port for receiving RTCP (and conventionally,
       for sending RTCP).

       By convention, UAs use their declared RTP listening ports as
       their RTP source ports as well.  The answer SDP will reach the
       remote UA, thus informing it of the address/port from which the
       MOH media will come, and presumably preventing the remote UA from
       ignoring the MOH media if the remote UA filters media packets
       based on the source address.  This functionality requires the SDP
       answer to contain the sending address in the c= line, even though
       the MOH source does not receive RTP.)

   6.  The executing UA sends this SDP answer as its SDP answer in the
       ACK for the re-INVITE to the remote UA.  The o= line in the
       answer must be modified to be within the sequence of o= lines
       previously generated by the executing UA in the dialog.  Any
       dynamic payload type number assignments that have been created in
       the answer must be recorded in the state of the original dialog.

   7.  Due to the sip.rendering feature parameter in the Contact of the
       re-INVITE and the media directionality in the SDP answer
       contained in the ACK, the on-hold state of the dialog is
       established (at the executing end).

   8.  After this point, the MOH source generates RTP containing the
       music-on-hold media, and sends it directly to the listening
       address/port of the remote UA.  The executing UA maintains two
       dialogs (one to the remote UA, one to the MOH source), but does
       not see or handle the MOH RTP.

2.2.  Taking a Call off Hold and Terminating the External Media Stream

   1.  The executing user instructs the executing UA to take the dialog
       off-hold.

   2.  The executing UA sends a re-INVITE to the remote UA with SDP that
       requests to receive media.  The Contact header of the re-INVITE
       does not include the '+sip.rendering="no"' field parameter.  (It
       may contain a sip.rendering field parameter with value "yes" or
       "unknown", or it may omit the field parameter.)  Thus this INVITE
       removes the on-hold state of the dialog (at the executing end).
       (Note that the version in o= line of the offered SDP must account
       for the SDP versions that were passed through from the MOH
       source.  Also note that any payload type numbers that were
       assigned in SDP provided by the MOH source must be respected.)





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   3.  When the remote UA sends a 2xx response to the re-INVITE, the
       executing UA sends a BYE request in the dialog to the MOH source.

   4.  After this point, the MOH source does not generate RTP and
       ordinary RTP flow is reestablished in the original dialog.

2.3.  Example Message Flow

   This section shows a message flow which is an example of this
   technique.  The scenario is: Alice establishes a call with Bob. Bob
   then places the call on hold, with music-on-hold provided from an
   external source.  Bob then takes the call off hold.  Note that this
   is just one possible message flow that illustrates this technique;
   numerous variations on these operations are allowed by the applicable
   standards.

   Alice             Bob       Music Source

   Alice establishes the call:

     |                |              |
     |    INVITE F1   |              |
     |--------------->|              |
     | 180 Ringing F2 |              |
     |<---------------|              |
     |    200 OK F3   |              |
     |<---------------|              |
     |     ACK F4     |              |
     |--------------->|              |
     |       RTP      |              |
     |<==============>|              |
     |                |              |

   Bob places Alice on hold, compelling Alice's UA to provide SDP:

     |                |              |
     |   INVITE F5    |              |
     |   (no SDP)     |              |
     |<---------------|              |
     |   200 OK F6    |              |
     |   (SDP offer)  |              |
     |--------------->|              |
     |                |              |

   Bob's UA initiates music-on-hold:

     |                |              |
     |                |  INVITE F7   |



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     |                |  (SDP offer, |
     |                |   rev. hold) |
     |                |------------->|
     |                | 200 OK F8    |
     |                | (SDP answer, |
     |                |  hold)       |
     |                |<-------------|
     |                |    ACK F9    |
     |                |------------->|
     |                |              |

   Bob's UA provides an SDP answer containing the address/port
   of the Music Source:

     |                |              |
     | ACK F10        |              |
     | (SDP answer,   |              |
     |  hold          |              |
     |<---------------|              |
     |    no RTP      |              |
     |<..............>|              |
     |     Music-on-hold RTP         |
     |<==============================|
     |                |              |

   The music on hold is active.

   Bob takes Alice off hold:

     |                |              |
     |  INVITE F11    |              |
     |  (SDP offer)   |              |
     |<---------------|              |
     |   200 OK F12   |              |
     |   (SDP answer) |              |
     |--------------->|              |
     |     ACK F13    |              |
     |<---------------|              |
     |                |    BYE F14   |
     |                |------------->|
     |                |    200 F15   |
     |                |<-------------|
     |       RTP      |              |
     |<==============>|              |
     |                |              |

   The normal media session between Alice and Bob is resumed.




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    /* Alice calls Bob. */

    F1 INVITE Alice -> Bob

    INVITE sips:bob@biloxi.example.com SIP/2.0
    Via: SIP/2.0/TLS atlanta.example.com:5061
     ;branch=z9hG4bK74bf9
    Max-Forwards: 70
    From: Alice <sips:alice@atlanta.example.com>;tag=1234567
    To: Bob <sips:bob@biloxi.example.com>
    Call-ID: 12345600@atlanta.example.com
    CSeq: 1 INVITE
    Contact: <sips:a8342043f@atlanta.example.com;gr>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces, gruu
    Content-Type: application/sdp
    Content-Length: [omitted]

    v=0
    o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
    s=
    c=IN IP4 atlanta.example.com
    t=0 0
    m=audio 49170 RTP/AVP 0
    a=rtpmap:0 PCMU/8000


    F2 180 Ringing Bob -> Alice

    SIP/2.0 180 Ringing
    Via: SIP/2.0/TLS atlanta.example.com:5061
     ;branch=z9hG4bK74bf9
     ;received=192.0.2.103
    From: Alice <sips:alice@atlanta.example.com>;tag=1234567
    To: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 1 INVITE
    Contact: <sips:bob@biloxi.example.com>
    Content-Length: 0


    F3 200 OK Bob -> Alice

    SIP/2.0 200 OK
    Via: SIP/2.0/TLS atlanta.example.com:5061
     ;branch=z9hG4bK74bf9
     ;received=192.0.2.103
    From: Alice <sips:alice@atlanta.example.com>;tag=1234567



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    To: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 1 INVITE
    Contact: <sips:bob@biloxi.example.com>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: [omitted]

    v=0
    o=bob 2890844527 2890844527 IN IP4 biloxi.example.com
    s=
    c=IN IP4 biloxi.example.com
    t=0 0
    m=audio 3456 RTP/AVP 0
    a=rtpmap:0 PCMU/8000


    F4 ACK Alice -> Bob

    ACK sips:bob@biloxi.example.com SIP/2.0
    Via: SIP/2.0/TLS atlanta.example.com:5061
     ;branch=z9hG4bK74bfd
    Max-Forwards: 70
    From: Alice <sips:alice@atlanta.example.com>;tag=1234567
    To: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 1 ACK
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces
    Content-Length: 0


    /* Bob places Alice on hold. */

    /* The re-INVITE contains no SDP, thus compelling Alice's UA
       to provide an offer. */

    F5 INVITE Bob -> Alice

    INVITE sips:a8342043f@atlanta.example.com;gr SIP/2.0
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bK874bk
    To: Alice <sips:alice@atlanta.example.com>;tag=1234567
    From: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 712 INVITE
    Contact: <sips:bob@biloxi.example.com>;+sip.rendering="no"



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    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces
    Content-Length: 0

    /* Alice's UA provides an SDP offer.
       Since it does not know that it is being put on hold,
       the offer is the same as the original offer and describes
       bidirectional media. */

    F6 200 OK Alice -> Bob

    SIP/2.0 200 OK
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bK874bk
     ;received=192.0.2.105
    To: Alice <sips:alice@atlanta.example.com>;tag=1234567
    From: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 712 INVITE
    Contact: <sips:a8342043f@atlanta.example.com;gr>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces, gruu
    Content-Type: application/sdp
    Content-Length: [omitted]

    v=0
    o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
    s=
    c=IN IP4 atlanta.example.com
    t=0 0
    m=audio 49170 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=active


    /* Bob's UA initiates music-on-hold. */

    /* This INVITE contains Alice's offer, but with the media
       direction set to "reverse hold", receive-only. */

    F7 INVITE Bob -> Music Source

    INVITE sips:music@source.example.com SIP/2.0
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bKnashds9
    Max-Forwards: 70
    From: Bob <sips:bob@biloxi.example.com>;tag=02134
    To: Music Source <sips:music@source.example.com>



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    Call-ID: 4802029847@biloxi.example.com
    CSeq: 1 INVITE
    Contact: <sips:bob@biloxi.example.com>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces, gruu
    Content-Type: application/sdp
    Content-Length: [omitted]

    v=0
    o=bob 2890844534 2890844534 IN IP4 atlanta.example.com
    s=
    c=IN IP4 atlanta.example.com
    t=0 0
    m=audio 49170 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=recvonly


    F8 200 OK Music Source -> Bob

    SIP/2.0 200 OK
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bKnashds9
     ;received=192.0.2.105
    From: Bob <sips:bob@biloxi.example.com>;tag=02134
    To: Music Source <sips:music@source.example.com>;tag=56323
    Call-ID: 4802029847@biloxi.example.com
    Contact: <sips:music@source.example.com>;automaton
         ;+sip.byeless;+sip.rendering="no"
    CSeq: 1 INVITE
    Content-Length: [omitted]

    v=0
    o=MusicSource 2890844576 2890844576 IN IP4 source.example.com
    s=
    c=IN IP4 source.example.com
    t=0 0
    m=audio 49170 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=sendonly


    F9 ACK Bob -> Music Source

    ACK sips:music@source.example.com SIP/2.0
    Via: SIP/2.0/TLS source.example.com:5061
     ;branch=z9hG4bK74bT6
    From: Bob <sips:bob@biloxi.example.com>;tag=02134



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    To: Music Source <sips:music@source.example.com>;tag=56323
    Max-Forwards: 70
    Call-ID: 4802029847@biloxi.example.com
    CSeq: 1 ACK
    Content-Length: 0


    /* Bob's UA now sends the ACK that completes the re-INVITE
       to Alice and completes the SDP offer/answer.
       The ACK contains the SDP received from the Music Source,
       and thus contains the address/port from which the Music Source
       will send media, and implies the address/port which the Music
       Source will use to send/receive RTCP. */

    F10 ACK Bob -> Alice

    ACK sips:a8342043f@atlanta.example.com;gr SIP/2.0
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bKq874b
    To: Alice <sips:alice@atlanta.example.com>;tag=1234567
    From: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 712 ACK
    Contact: <sips:bob@biloxi.example.com>;+sip.rendering="no"
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces
    Content-Length: [omitted]

    v=0
    o=bob 2890844527 2890844528 IN IP4 biloxi.example.com
    s=
    c=IN IP4 source.example.com
    t=0 0
    m=audio 49170 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=sendonly

    /* Bob picks up the call by sending a re-INVITE to Alice. */

    F11 INVITE Bob -> Alice

    INVITE sips:a8342043f@atlanta.example.com;gr SIP/2.0
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bK874bk
    To: Alice <sips:alice@atlanta.example.com>;tag=1234567
    From: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 713 INVITE



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    Contact: <sips:bob@biloxi.example.com>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: [omitted]

    v=0
    o=bob 2890844527 2890844529 IN IP4 biloxi.example.com
    s=
    c=IN IP4 biloxi.example.com
    t=0 0
    m=audio 3456 RTP/AVP 0
    a=rtpmap:0 PCMU/8000


    F12 200 OK Alice -> Bob

    SIP/2.0 200 OK
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bK874bk
     ;received=192.0.2.105
    To: Alice <sips:alice@atlanta.example.com>;tag=1234567
    From: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 713 INVITE
    Contact: <sips:a8342043f@atlanta.example.com;gr>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces, gruu
    Content-Type: application/sdp
    Content-Length: [omitted]

    v=0
    o=alice 2890844526 2890844527 IN IP4 atlanta.example.com
    s=
    c=IN IP4 atlanta.example.com
    t=0 0
    m=audio 49170 RTP/AVP 0
    a=rtpmap:0 PCMU/8000


    F13 ACK Bob -> Alice

    ACK sips:a8342043f@atlanta.example.com;gr SIP/2.0
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bKq874b
    To: Alice <sips:alice@atlanta.example.com>;tag=1234567
    From: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com



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    CSeq: 713 ACK
    Contact: <sips:bob@biloxi.example.com>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces
    Content-Length: 0


    F14 BYE Bob -> Music Source

    BYE sips:music@source.example.com SIP/2.0
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bK74rf
    Max-Forwards: 70
    From: Bob <sips:bob@biloxi.example.com>;tag=02134
    To: Music Source <sips:music@source.example.com>;tag=56323
    Call-ID: 4802029847@biloxi.example.com
    CSeq: 2 BYE
    Contact: <sips:bob@biloxi.example.com>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces, gruu
    Content-Length: [omitted]


    F15 200 OK Music Source -> Bob

    SIP/2.0 200 OK
    Via: SIP/2.0/TLS atlanta.example.com:5061
     ;branch=z9hG4bK74rf
     ;received=192.0.2.103
    From: Bob <sips:bob@biloxi.example.com>;tag=02134
    To: Music Source <sips:music@source.example.com>;tag=56323
    Call-ID: 4802029847@biloxi.example.com
    Contact: <sips:music@source.example.com>;automaton
         ;+sip.byeless;+sip.rendering="no"
    CSeq: 2 BYE
    Content-Length: 0


    /* Normal media session between Alice and Bob is resumed */

2.4.  Re-INVITE and UPDATE from the Remote UA

   While the call is on-hold, the remote UA can send a request to modify
   the SDP or the feature parameters of its Contact header.  This can be
   done with either an INVITE or UPDATE method, both of which have much
   the same effect in regard to MOH.

   A common reason for a re-INVITE is when the remote UA desires to put



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   the dialog on hold on its end.  And because of the need to support
   this case, an implementation must process INVITEs and UPDATEs during
   the on-hold state as described below.

   The executing UA handles these requests by echoing requests and
   responses: an incoming request from the remote UA causes the
   executing UA to send a similar request to the MOH source and an
   incoming response from the MOH source causes the executing UA to send
   a similar response to the remote UA.  In all cases, SDP offers or
   answers that are received are added as bodies to the stimulated
   request or response to the other UA.

   The passed-through SDP will usually need its o= line modified.  The
   directionality attributes may need to be restricted.  In regard to
   payload type numbers, since the mapping has already been established
   within the MOH dialog, a=rtpmap lines need not be added.

2.5.  INVITE with Replaces

   The executing UA must be prepared to receive INVITE requests with
   Replaces headers that replace the original dialog, and similarly it
   must be prepared to receive REFER requests within the dialog.  The
   SDP within the new dialog is negotiated by being passed through to
   the MOH source within a new dialog with the MOH source.  The SDP
   offer or answer can be passed to the MOH source with only
   modification to the o= line and directionality attributes.

   In some cases, the previous dialog with the MOH source can be reused,
   but only if the executing UA presents the first offer within the new
   dialog, as otherwise there is no way to force the RTP payload types
   that have been used previously in the MOH dialog to be mapped to the
   correct codecs in the new dialog.

2.6.  Re-INVITE and UPDATE from the Music-On-Hold Source

   It is possible for the MOH source to send an INVITE or UPDATE
   request, and the executing UA can support doing so in similar manner
   as requests from the remote UA.  However, if the MOH source is within
   the same administrative domain as the executing UA, the executing UA
   may have knowledge that the MOH source will not (or need not) make
   such requests, and so can respond to any such request with a failure
   response, avoiding the need to pass the request through.

   However, in an environment in which ICE[ice] is supported, the MOH
   source may need to send requests as part of ICE negotiation with the
   remote UA.  Hence, in environments that support ICE, the executing UA
   must be able to pass through requests from the MOH source as well as
   requests from the remote UA.



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   Again, as SDP is passed through, its o= line will need to be
   modified.  In some cases, the directionality attributes will need to
   be restricted.

2.7.  Handling Payload Type Numbers

2.7.1.  Analysis

   In this technique, the MOH source generates an SDP answer that the
   executing UA presents to the remote UA as an answer within the
   original dialog.  In basic functionality, this presents no problem,
   because [offer-answer] (section 6.1, at the very end) specifies that
   the payload type numbers used in either direction of RTP are the ones
   specified in the SDP sent by the recipient of the RTP.  Thus, the MOH
   source will send RTP to the remote UA using the payload type numbers
   specified in the offer SDP it received (ultimately) from the remote
   UA.

   But strict compliance to [offer-answer] (section 8.3.2) requires that
   payload type numbers used in SDP may only duplicate the payload type
   numbers used in any previous SDP sent in the same direction if the
   payload type numbers represent the same media format (codec) as they
   did previously.  However, the MOH source has no knowledge of the
   payload type numbers previously used in the original dialog, and it
   may accidentally specify a different media format for a previously
   used payload type number in its answer (or in a subsequently
   generated INVITE or UPDATE).  This would cause no problem with media
   decoding, as it cannot send any format that was not in the remote
   UA's offer, but it would violate [offer-answer].

   Strictly speaking, it is impossible to avoid this problem because the
   generator of a first answer in its dialog can choose the payload
   numbers independently of the payload numbers in the offer, and the
   MOH server believes that its answer is first in the dialog.  Thus the
   only absolute solution is to have the executing UA rewrite the SDP
   that passes through it to reassign payload type numbers, which would
   also require it to rewrite the payload type numbers in the RTP
   packets -- a very undesirable solution.

   The difficulty solving this problem (and similar problems in other
   situations) argues that strict adherence should not be required to
   the rule that payload type numbers not be reused for different
   codecs.

   The remainder of this section is devoted to describing a technique to
   eliminate this problem, in case it is of practical significance in
   some application.  We do not expect that user agents would need to
   implement it in most applications.



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2.7.2.  Solution to the Problem

   However, we can construct a technique that will strictly adhere to
   the payload type rule by exploiting a SHOULD-level requirement in
   [offer-answer] (section 6.1): "In the case of RTP, if a particular
   codec was referenced with a specific payload type number in the
   offer, that same payload type number SHOULD be used for that codec in
   the answer."  Or rather, we exploit the "implied requirement" that if
   a specific payload number in the offer is used for a particular
   codec, then the answer should not use that payload number for a
   different codec.  If the MOH source obeys this restriction, the
   executing UA can modify the offer SDP to "reserve" all payload type
   numbers that have ever been offered by the executing UA to prevent
   the MOH source from using them for different media formats.

   When the executing UA is composing the INVITE to the MOH source, it
   compiles a list of all the (dynamically-assigned) payload type
   numbers and associated media formats which have been used by it (or
   by MOH sources on its behalf) in the original dialog.  (The executing
   UA must be maintaining a list of all previously used payload type
   numbers anyway, in order to comply with [offer-answer].)

   Any payload type number that is present in the offer but has been
   used previously by the executing UA in the original dialog for a
   different media format is rewritten to describe a dummy media format.
   A payload type number is added to describe the deleted media format,
   the number being either previously unused or previously used by the
   executing UA for that media format.

   Any further payload type numbers which have been used by the
   executing UA in the original dialog but which are not mapped to a
   media format in the current offer are then mapped to a dummy media
   format.

   The result is that the modified offer SDP:

   1.  offers the same set of media formats (ignoring dummies) as the
       original offer SDP (though possibly with different payload type
       numbers),

   2.  associates every payload type number either with a dummy media
       format or with the media format that the executing UA has
       previously used it for, and

   3.  provides a (real or dummy) media format for every payload type
       number that the executing UA has previously used.

   These properties are sufficient to force an MOH server that obeys the



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   implied requirement to generate an answer that is a correct answer to
   the original offer and is also compatible with previous SDP from the
   executing UA.

   Note that any re-INVITEs from the remote UA that the executing UA
   passes through to the MOH server require similar modification, as
   payload type numbers that the MOH server receives in past offers are
   not absolutely reserved against its use (as they have not been sent
   in SDP by the MOH server) nor is there a SHOULD-level proscription
   against using them in the current answer (as they do not appear in
   the current offer).

   This should provide an adequate solution to the problems with payload
   type numbers, as it will fail only if (1) the remote UA is particular
   that other UAs follow the rule about not redefining payload type
   numbers, and (2) the MOH server does not follow the implied
   requirement of [offer-answer] section 6.1.

2.7.3.  Example of the Solution

   Let us show how this process works by modifying the example of
   Section 2.3 with this specific assignment of supported codecs:

      Alice supports formats X and Y

      Bob supports formats X and Z

      Music Source supports formats Y and Z

   In this case, the SDP exchanges are:

      F1 offers X and Y, F3 answers X and Z (which cannot be used)

      F6 offers X and Y, but F7 offers X, Y, and a place-holder to block
      use of type 92

      F8/F10 answers Y

   The messages that are changed from Section 2.3 are:

    F1 INVITE Alice -> Bob

    INVITE sips:bob@biloxi.example.com SIP/2.0
    Via: SIP/2.0/TLS atlanta.example.com:5061
     ;branch=z9hG4bK74bf9
    Max-Forwards: 70
    From: Alice <sips:alice@atlanta.example.com>;tag=1234567
    To: Bob <sips:bob@biloxi.example.com>



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    Call-ID: 12345600@atlanta.example.com
    CSeq: 1 INVITE
    Contact: <sips:a8342043f@atlanta.example.com;gr>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces, gruu
    Content-Type: application/sdp
    Content-Length: [omitted]

    v=0
    o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
    s=
    c=IN IP4 atlanta.example.com
    t=0 0
    m=audio 49170 RTP/AVP 90 91
    a=rtpmap:90 X/8000
    a=rtpmap:91 Y/8000


    F3 200 OK Bob -> Alice

    SIP/2.0 200 OK
    Via: SIP/2.0/TLS atlanta.example.com:5061
     ;branch=z9hG4bK74bf9
     ;received=192.0.2.103
    From: Alice <sips:alice@atlanta.example.com>;tag=1234567
    To: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 1 INVITE
    Contact: <sips:bob@biloxi.example.com>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: [omitted]

    v=0
    o=bob 2890844527 2890844527 IN IP4 biloxi.example.com
    s=
    c=IN IP4 biloxi.example.com
    t=0 0
    m=audio 3456 RTP/AVP 90 92
    a=rtpmap:90 X/8000
    a=rtpmap:92 Z/8000


    F6 200 OK Alice -> Bob

    SIP/2.0 200 OK
    Via: SIP/2.0/TLS biloxi.example.com:5061



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     ;branch=z9hG4bK874bk
     ;received=192.0.2.105
    To: Alice <sips:alice@atlanta.example.com>;tag=1234567
    From: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 712 INVITE
    Contact: <sips:a8342043f@atlanta.example.com;gr>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces, gruu
    Content-Type: application/sdp
    Content-Length: [omitted]

    v=0
    o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
    s=
    c=IN IP4 atlanta.example.com
    t=0 0
    m=audio 49170 RTP/AVP 90 91
    a=rtpmap:90 X/8000
    a=rtpmap:91 Y/8000
    a=active


    F7 INVITE Bob -> Music Source

    INVITE sips:music@source.example.com SIP/2.0
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bKnashds9
    Max-Forwards: 70
    From: Bob <sips:bob@biloxi.example.com>;tag=02134
    To: Music Source <sips:music@source.example.com>
    Call-ID: 4802029847@biloxi.example.com
    CSeq: 1 INVITE
    Contact: <sips:bob@biloxi.example.com>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
    Supported: replaces, gruu
    Content-Type: application/sdp
    Content-Length: [omitted]

    v=0
    o=bob 2890844534 2890844534 IN IP4 atlanta.example.com
    s=
    c=IN IP4 atlanta.example.com
    t=0 0
    m=audio 49170 RTP/AVP 90 91 92
    a=rtpmap:90 X/8000
    a=rtpmap:91 Y/8000
    a=rtpmap:92 x-reserved/8000



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    a=recvonly


    F8 200 OK Music Source -> Bob

    SIP/2.0 200 OK
    Via: SIP/2.0/TLS biloxi.example.com:5061
     ;branch=z9hG4bKnashds9
     ;received=192.0.2.105
    From: Bob <sips:bob@biloxi.example.com>;tag=02134
    To: Music Source <sips:music@source.example.com>;tag=56323
    Call-ID: 4802029847@biloxi.example.com
    Contact: <sips:music@source.example.com>;automaton
         ;+sip.byeless;+sip.rendering="no"
    CSeq: 1 INVITE
    Content-Length: [omitted]

    v=0
    o=MusicSource 2890844576 2890844576 IN IP4 source.example.com
    s=
    c=IN IP4 source.example.com
    t=0 0
    m=audio 49170 RTP/AVP 91
    a=rtpmap:91 Y/8000
    a=sendonly

2.8.  Dialog/Session Timers

   The executing UA may discover that either the remote UA or the MOH
   source wishes to use dialog/session liveness timers.[timers] Since
   the timers verify the liveness of dialogs, not sessions (despite the
   terminology of [timers]), the executing UA can support the timers on
   each dialog (to the remote UA and to the MOH source) independently.
   (If the executing UA becomes obliged to initiate a refresh
   transaction, it must send an offerless UPDATE or re-INVITE, as if it
   sends an offer, the remote element has the opportunity to provide an
   answer which is different from its previous SDP, which could not
   easily be conveyed to the other remote element.)













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3.  Advantages

   This technique for providing music-on-hold has advantages over other
   methods now in use:

   1.  The original dialog is not transferred to another UA, so the
       "remote endpoint URI" displayed by the remote endpoint's user
       interface and dialog event package[dialog-event] does not change
       during the call, as contrasted to the method in
       [service-examples] section 2.3.  This URI is usually displayed to
       the user as the the name and number of the other party on the
       call, and it is desirable for it not to change to that of the MOH
       server.

   2.  Compared to [service-examples], this method does not require use
       of an out-of-dialog REFER, which is not otherwise used much in
       SIP.  Out-of-dialog REFERs may not be routed correctly, since
       neither the From nor Contact URI of the original dialog may route
       correctly to the remote UA.  Also, out-of-dialog requests to UA
       URIs may not be handled correctly by authorization mechanisms.

   3.  The music-on-hold media are sent directly from the music-on-hold
       source to the remote UA, rather than being relayed through the
       executing UA.  This reduces the computational load on the
       executing UA and can reduce the load on the network (by
       eliminating "hairpinning" of the media through the link serving
       the executing UA).

   4.  The remote UA sees, in the incoming SDP, the address/port that
       the MOH source will send MOH media from (assuming that the MOH
       source follows the convention of sending its media from its
       advertised media listening address/port).  Thus the remote UA
       will render the MOH media even if is filtering incoming media
       based on originating address as a media security measure.

   5.  The technique requires relatively simple manipulation of SDP, and
       in particular: (1) does not require a SIP element to modify
       unrelated SDP to be acceptable to be sent within an already
       established sequence of SDP (a problem with [service-examples-11]
       section 2.3), and (2) does not require converting an SDP answer
       into an SDP offer (which was a problem with the -00 version of
       this document, as well as with [service-examples-11]).









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4.  Caveats

4.1.  Offering All Available Media Formats

   Unnecessary failures can happen if SDP offerers do not always offer
   all media formats that they support.  Doing so is considered best
   practice ([offer-answer-bis] section 5.1 and 5.3), but some elements
   offer only formats that have already been in use in the dialog.

   An example of how omitting media formats in an offer can lead to
   failure is as follows: Suppose that the UAs in Section 2.3 each
   support the following media formats:

      Alice supports formats X and Y

      Bob supports formats X and Z

      Music Source supports formats Y and Z

   In this case, the SDP exchanges are:

   1.  Alice calls Bob:
       Alice offers X and Y (message F1)
       Bob answers X (F3)

   2.  Bob puts Alice on hold:
       Alice (via Bob) offers X and Y (F6 and F7)
       Music Source (via Bob) answers Y (F8 and F10)

   3.  Bob takes Alice off hold:
       Bob offers X and Z (F11)
       Alice answers X (F12)

   Note that in exchange 2, if Alice assumes that because only format X
   is in use that she should offer only X, the exchange fails.  In
   exchange 3, Bob offers formats X and Z, even though neither is in use
   at the time (because Bob is not involved in the media streams).

4.2.  Handling re-INVITES in a B2BUA

   Many UAs provide MOH in the interval during which it is processing a
   blind transfer, between receiving the REFER and receiving the final
   response to the stimulated INVITE.  This process involves switching
   the user's interface between three media sources: (1) the session of
   the original dialog, (2) the session with the MOH server, and (3) the
   session of the new dialog, and involves a number of race conditions
   that must be handled correctly.  If the UA is a B2BUA whose "other
   side" is maintaining a single dialog with another UA, each switching



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   of media sources potentially causes a re-INVITE transaction within
   the other-side dialog.  Since re-INVITEs take time and must be
   sequenced correctly ([sip] section 14), such a B2BUA must allow the
   events on each side to be non-synchronous and must coordinate them
   correctly.  Failing to do so will lead to "glare" errors (491 or
   500), leaving the other-side UA not rendering the correct session.













































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5.  Security Considerations

   Some UAs filter incoming media based on the address of origin as a
   media security measure.  The technique described in this document
   ensures that any UA that should render MOH media will be informed of
   the source address of the media via the SDP that it receives.  This
   should allow such UAs to filter without interfering with MOH
   operation.











































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6.  Acknowledgments

   The original version of this proposal was derived from
   [service-examples-11] section 2.3 and the similar implementation of
   MOH in the Snom UA.  Significant improvements to the sequence of
   operations, allowing improvements to the SDP handling, were suggested
   by Venkatesh[venkatesh].

   John Elwell[elwell] pointed out the need for the executing UA to pass
   through re-INVITEs/UPDATEs in order to allow ICE negotiation.

   Paul Kyzivat[kyzivat-1][kyzivat-2] pointed out the difficulties
   regarding reuse of payload type numbers and considerations that could
   be used to avoid those difficulties, leading to the writing of
   Section 2.7.

   Paul Kyzivat suggested adding Section 4.1 showing why offerers should
   always include all supported formats.

   M. Ranganathan pointed out the difficulties experienced by a B2BUA
   (Section 4.2) due to the multiple changes of media source.

   Section 4.1 was significantly clarified based on advice from Attila
   Sipos[sipos].

   The need to discuss dialog/session timers[Section 2.8] was pointed
   out by Rifaat Shekh-Yusef[shekh-yusef].

   Robert Sparks clarified the purpose of the "Best Common Practice"
   status, leading to revising the intended status of this document to
   "Informational".Section 1.1




















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7.  Revision History

   (Note to RFC Editor: Please remove this entire section upon
   publication as an RFC.

7.1.  Changes from draft-worley-service-example-00 to
      draft-worley-service-example-01

   Removed the original "Example Message Flow" and promoted the
   "Alternative Example Message Flow" to replace it because of a number
   of flaws that were found during the discussion of -00 on the SIPPING
   mailing list.

   Described the use of the sip.rendering feature parameter to indicate
   on-hold status.

7.2.  Changes from draft-worley-service-example-01 to
      draft-worley-service-example-02

   Added discussion of passing though re-INVITEs and UPDATEs.

   Added discussion of payload type numbers.

   Added Acknowledgments section.

7.3.  Changes from draft-worley-service-example-02 to
      draft-worley-service-example-03

   Added Section 4.1 showing the importance of the offerer always
   including all supported media formats.

   Updated references.

   Revised handling of payload type numbers when passing offer to MOH
   server Section 2.7, based on observations by Paul Kyzivat.

7.4.  Changes from draft-worley-service-example-03 to
      draft-worley-service-example-04

   Added Section 4.2 discussing handling of re-INVITEs by B2BUAs when
   using this method.

   Added "avoidance of out-of-dialog REFER" as an advantage.Section 3

   Added "automaton", "sip.rendering", and "sip.byeless" feature tags to
   the Contact URI of the Music Server in the
   examples.[dialog-event][ua-capabilities]




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   Added initial discussion of dialog/session timer support.Section 2.8

   Revised handling of payload type numbers based on further
   observations by Paul Kyzivat[kyzivat-2].

7.5.  Changes from draft-worley-service-example-04 to
      draft-worley-service-example-05

   Changed references to "SPIT" to refer to "media security", per
   suggestion by Scott Lawrence.

   Removed reference to the idea of having the executing UA not maintain
   session timers itself, but rather, passing through session timer
   negotiation and updates.  Examination showed this idea to be much
   more complex to implement than having the executing UA terminate
   session timers itself for both dialogs.  (Suggested by Rifaat Shekh-
   Yusef.)

   On advice from Robert Sparks, changed the "intended status" from
   "BCP" to "Informational", and added a section to explain the change.

   Noted that the rule on not reusing payload type numbers is
   undesirable because it complicates third-party operations (as noted
   by Paul Kyzivat[kyzivat-3]).

7.6.  Changes from draft-worley-service-example-05 to
      draft-worley-service-example-06

   Updated author's contact information.

   On suggestion from John Elwell, added mention that the Music Source's
   SDP address/port implies its RTCP address/port, which will be used to
   receive RTCP.

   Updated references to [service-examples] and [service-examples-11] to
   specify the sections of documents, which are the ones that discuss
   music on hold.














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8.  References

8.1.  Normative References

   [offer-answer]
              Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with the Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [sdp]      Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [sip]      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.

   [timers]   Donovan, S. and J. Rosenberg, "Session Timers in the
              Session Initiation Protocol (SIP)", RFC 4028, April 2005.

8.2.  Informative References

   [dialog-event]
              Rosenberg, J., Schulzrinne, H., and R. Mahy, "An INVITE-
              Initiated Dialog Event Package for the Session Initiation
              Protocol (SIP)", RFC 4235, November 2005.

   [elwell]   Elwell, J., "Subject: [Sipping] RE: I-D
              Action:draft-worley-service-example-00.txt", IETF Sipping
              mailing list msg14678, November 2007.

   [ice]      Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [kyzivat-1]
              Kyzivat, P., "Subject: Re: [Sipping] I-D
              ACTION:draft-ietf-sipping-service-examples-11.txt", IETF
              Sipping mailing list msg12181, October 2006.

   [kyzivat-2]
              Kyzivat, P., "[Sip-implementors]
              draft-worley-service-example-02", sip-implementors mailing
              list 020426, September 2008.

   [kyzivat-3]
              Kyzivat, P., "[Sip-implementors]



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              draft-worley-service-example-02", sip-implementors mailing
              list 020387, September 2008.

   [offer-answer-bis]
              Okumura, S. and P. Kyzivat, "SIP (Session Initiation
              Protocol) Usage of the Offer/Answer Model",
              I-D draft-ietf-sipping-sip-offeranswer-13, May 2010.

   [service-examples]
              Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and
              K. Summers, "Session Initiation Protocol Service
              Examples", RFC 5359, October 200.

   [service-examples-11]
              Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and
              K. Summers, "Session Initiation Protocol Service
              Examples", I-D draft-ietf-sipping-service-examples-11,
              October 2006.

   [shekh-yusef]
              Rifaat Shekh-Yusef, "[sipcore]
              draft-worley-service-example-03", IETF Sipcore mailing
              list msg00580, July 2009.

   [sipos]    Attila Sipos, "RE: [Sip-implementors]
              draft-worley-service-example-02", sip-implementors mailing
              list 022002, March 2009.

   [ua-capabilities]
              Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
              "Indicating User Agent Capabilities in the Session
              Initiation Protocol (SIP)", RFC 3840, August 2004.

   [venkatesh]
              Venkatesh, "Subject: Re: [Sipping] I-D
              ACTION:draft-ietf-sipping-service-examples-11.txt", IETF
              Sipping mailing list msg12180, October 2006.














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Author's Address

   Dale R. Worley
   Avaya Inc.
   600 Technology Park Dr.
   Billerica, MA  01821
   US

   Email: dworley@avaya.com
   URI:   http://www.avaya.com









































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