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SIP                                                          D.R. Worley
Internet-Draft                                                Bluesocket
Expires: July 19, 2008                                  January 16, 2008


      Session Initiation Protocol Service Example -- Music on Hold
                    draft-worley-service-example-01

Status of this Memo

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   applicable patent or other IPR claims of which he or she is aware
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   This Internet-Draft will expire on July 19, 2008.

Copyright Notice

   Copyright (C) The IETF Trust (2008).

Abstract

   The "music on hold" feature is one of the most desired features of
   telephone systems in the business environment.  "Music on hold" is
   when one party to a call has the call "on hold", the other party
   receives a media stream (often either music or advertising).
   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, but is simpler than the methods
   previously documented.



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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Technique  . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.1.  Placing a Call on Hold and Providing an External Media
           Stream . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.2.  Taking a Call off Hold and Terminating the External
           Media Stream . . . . . . . . . . . . . . . . . . . . . . .  5
     2.3.  Example Message Flow . . . . . . . . . . . . . . . . . . .  6
     2.4.  Managing o= Lines  . . . . . . . . . . . . . . . . . . . . 14
   3.  Advantages . . . . . . . . . . . . . . . . . . . . . . . . . . 16
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 17
   5.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 18
   6.  Revision History . . . . . . . . . . . . . . . . . . . . . . . 19
     6.1.  Changes from draft-worley-service-example-00 to
           draft-worley-service-example-01  . . . . . . . . . . . . . 19
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 20
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 20
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21
   Intellectual Property and Copyright Statements . . . . . . . . . . 22






























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1.  Introduction

   Within SIP[1]-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 when one party to a call has the call "on
   hold", that party's telephone provides a media stream to be rendered
   to 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.






































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2.  Technique

   The essence of the technique is that when the user's UA (referred to
   hereafter as the "executing UA") establishes the hold state, it uses
   third party call control mechanisms to direct RTP from an audio
   source service to the held party.  The executing UA establishes a
   dialog with the service to negotiate the audio stream, but does not
   act as a media relay; media flows from the service to the held party.

   In order to accomplish this within the offer/answer model, the
   executing UA sends a re-INVITE to the remote UA to establish the hold
   state, but in that INVITE it provides no SDP offer, thus compelling
   the remote UA to provide an SDP offer.  The executing UA then uses
   that offer (modified as described in Section 2.4) 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 Providing 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 SHOULD include the
       '+sip.rendering="no"' field parameter so that the intent is clear
       and so that any dialog events indicate that it is putting the
       call on hold.[6] (The sip.rendering feature parameter is not
       required by the MOH mechanics, but rather to facilitate reporting
       the on-hold status in dialog events.)

   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
       perhaps "inactive".  But the remote UA may offer to send media.

   4.  The executing UA uses this offer to construct the offer SDP of an
       initial INVITE that it sends to the configured music-on-hold
       (MOH) source.  The SDP in this request is, with the exceptions
       below, copied from the SDP returned by the remote UA in the
       previous step, particularly regarding the provided listening
       address/port and payload type numbers.  The changes to the SDP
       offer are:




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       *  The o= line of the offer is modified according to Section 2.4.

       *  Any explicit or implicit media directionality of "sendrecv" is
          replaced with "recvonly", and any media directionality of
          "sendonly" is replaced with "inactive".

   5.  The MOH 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[2] specify
       "sendonly" or "inactive" as the directionality for all media
       streams.  (Although this address/port should receive no RTP, the
       answer SDP will reach the remote UA, informing it of the address/
       port from which the MOH media will come, and presumably
       preventing it from ignoring the MOH media as SPIT.  This requires
       the SDP answer to provide a real address/port in the c= line,
       even though the MOH server does not receive RTP.)

   6.  The executing UA copies the MOH service SDP answer, modifying it
       as listed below, and sends it as its SDP answer in the ACK for
       the re-INVITE to the remote UA.

       *  The o= line of the answer is modified according to
          Section 2.4.

   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 (at the executing end) of
       the dialog is established.  (The sip.rendering feature parameter
       will be reported in the dialog events generated by both UAs,
       allowing any monitoring application to determine that the call is
       on hold.)

   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



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       removes the on-hold state (at the executing end) of the dialog.

   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
       to terminate the MOH media stream.

   4.  After this point, the MOH source does not generate RTP and
       ordinary RTP flow is re-established 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 server.  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 Server

   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)  |              |
     |--------------->|              |
     |                |              |




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   Bob's UA initiates music-on-hold:

     |                |              |
     |                |  INVITE F7   |
     |                |  (SDP offer, |
     |                |   rev. hold) |
     |                |------------->|
     |                | 200 OK F8    |
     |                | (SDP answer, |
     |                |  hold)       |
     |                |<-------------|
     |                |    ACK F9    |
     |                |------------->|
     |                |              |

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

     |                |              |
     | ACK (hold) F10 |              |
     | (SDP answer)   |              |
     |<---------------|              |
     |    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      |              |
     |<==============>|              |
     |                |              |



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   The normal media session between Alice and Bob is resumed.

   Message Details

    /* 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



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    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 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



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    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"
    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 Server

    INVITE sips:music@server.example.com SIP/2.0
    Via: SIP/2.0/TLS biloxi.example.com:5061



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     ;branch=z9hG4bKnashds9
    Max-Forwards: 70
    From: Bob <sips:bob@biloxi.example.com>;tag=02134
    To: Music Server <sips:music@server.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 0
    a=rtpmap:0 PCMU/8000
    a=recvonly


    F8 200 OK Music Server -> 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 Server <sips:music@server.example.com>;tag=56323
    Call-ID: 4802029847@biloxi.example.com
    Contact: <sips:music@server.example.com>
    CSeq: 1 INVITE
    Content-Length: [omitted]

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


    F9 ACK Bob -> Music Server

    ACK sips:music@server.example.com SIP/2.0



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    Via: SIP/2.0/TLS server.example.com:5061
     ;branch=z9hG4bK74bT6
    From: Bob <sips:bob@biloxi.example.com>;tag=02134
    To: Music Server <sips:music@server.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 Server,
       and thus contains the address/port from which the Music Server
       will send media. */

    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 server.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



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    Call-ID: 12345600@atlanta.example.com
    CSeq: 713 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 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



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    From: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    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 Server

    BYE sips:music@server.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 Server <sips:music@server.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 Server -> Alice

    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 Server <sips:music@server.example.com>;tag=56323
    Call-ID: 4802029847@biloxi.example.com
    CSeq: 2 BYE
    Content-Length: 0


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

2.4.  Managing o= Lines

   The rulesSection 2.4 regarding the use of o= lines in successive SDP
   offers/answers during a dialog are quit strict.  In particular,
   successive SDPs sent by one UA in the dialog must have identical o=
   lines, other than that the version number field must be incremented
   by 1.  The single exception is that if two or more successive SDPs
   are exactly the same, they may have the same version number.



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   In order for the executing UA to conform to these rules, it must
   modify the o= lines in any SDP that it passes from one dialog to the
   other.  In particular:

   1.  The SDP in the initial INVITE to to the MOH server must have its
       o= line replaced with a newly constructed o= line that is valid
       within the dialog with the MOH server -- See [3] for the details.

   2.  Within any later SDP copied from the original dialog to the MOH
       dialog, the o= line must be replaced with a copy of the o= line
       that the executing UA used in the previous SDP that it sent in
       the MOH dialog, with the version number incremented by 1.

   3.  Within any SDP copied from the MOH dialog to the original dialog,
       the o= line must be replaced with a copy of the o= line that the
       executing UA used in the previous SDP that it sent in the
       original dialog, with the version number incremented by 1.

   4.  When the executing UA takes the call off-hold, the o= line of the
       SDP that it sends in the re-INVITE must have a version number one
       greater than in the o= line of the last SDP the executing UA
       copied from the MOH dialog.





























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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[6] does not change during the
       call.[4]

   2.  The music-on-hold media is sent directly from the music-on-hold
       source to the remote UA, rather than being relayed through the
       holding UA.

   3.  The remote UA sees, in the incoming SDP, the address/port that
       the MOH server will send MOH media from, thus allowing it to
       render the media, even if it is filtering incoming media based on
       originating address as a SPIT preventative.

   4.  The technique requires relatively simple manipulation of SDP, and
       in particular: (1) does not require a SIP agent to modify
       unrelated SDP to be acceptable to be sent within an already
       established sequence of SDP (a problem with [5]), 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
       [5]).

























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4.  Security Considerations

   Some UAs filter incoming media based on the address of origin in
   order to avoid SPIT.  This technique ensures that any UA that should
   render MOH media will be informed of the source address via the SDP
   that it receives.  This should allow such UAs to filter without
   interfering with MOH operation.












































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5.  Acknowledgments

   The original version of this proposal was derived from [5] and the
   similar implementation of MOH in the Snom UA.  Significant
   improvements to sequence of operations, allowing improvements to the
   SDP handling, were suggested by Venkatesh[7].

   John Elwell[8] pointed out the need for the executing UA to pass
   through re-INVITEs/UPDATEs in order to allow ICE negotiation, which
   suggested to me the need for pass-through to handle the remote UA
   placing its end of the call on-hold.

   This version benefitted from Scott Lawrence's careful reading and
   comments.





































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6.  Revision History

6.1.  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 described in the discussion of -00 on the SIPPING
   mailing list.

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

   Added Acknowledgments section.

   Added separate section on the management of o= lines.



































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7.  References

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

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

7.2.  Informative References

   [4]  Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and K.
        Summers, "Session Initiation Protocol Service Examples",
        I-D draft-ietf-sipping-service-examples-13, July 2007.

   [5]  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.

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

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

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
















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Author's Address

   Dale R. Worley
   Bluesocket Inc.
   10 North Ave.
   Burlington, MA  01803
   US

   Phone: +1 781 229 0533 x173
   Email: dworley@pingtel.com
   URI:   http://www.pingtel.com








































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