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SIP                                                            D. Worley
Internet-Draft                                                    Nortel
Expires: March 1, 2009                                   August 28, 2008


      Session Initiation Protocol Service Example -- Music on Hold
                    draft-worley-service-example-02

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   This Internet-Draft will expire on March 1, 2009.

Copyright Notice

   Copyright (C) The IETF Trust (2008).















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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, but is simpler
   than the methods previously documented.


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.  Re-INVITE and UPDATE from the Remote UA  . . . . . . . . . 14
     2.5.  INVITE with Replaces . . . . . . . . . . . . . . . . . . . 15
     2.6.  Re-INVITE and UPDATE from the Music-On-Hold Source . . . . 15
     2.7.  Payload Type Numbers . . . . . . . . . . . . . . . . . . . 15
   3.  Advantages . . . . . . . . . . . . . . . . . . . . . . . . . . 17
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 18
   5.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 19
   6.  Revision History . . . . . . . . . . . . . . . . . . . . . . . 20
     6.1.  Changes from draft-worley-service-example-00 to
           draft-worley-service-example-01  . . . . . . . . . . . . . 20
     6.2.  Changes from draft-worley-service-example-01 to
           draft-worley-service-example-02  . . . . . . . . . . . . . 20
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 21
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 21
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 22
   Intellectual Property and Copyright Statements . . . . . . . . . . 23












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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 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.






































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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 to establish the
   hold state, it provides no SDP[3] offer[2][11], 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 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 includes the
       '+sip.rendering="no"' field parameter to indicate that it is
       putting the call on hold.[6]

   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 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[2] specify
       "sendonly" or "inactive" as the directionality for all media
       streams.  (Although this address/port should receive no RTP, by
       convention UAs use their declared RTP listening ports as their
       RTP source ports as well.  The answer SDP will reach the remote



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       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 as SPIT.  This functionality requires the
       SDP answer to contain the sending address/port 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, and that any payload type numbers that were assigned in
       SDP provided by the MOH source must be respected.)

   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 re-established in the original dialog.






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



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     |                |  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.

   Message Details

    /* Alice calls Bob. */

    F1 INVITE Alice -> Bob



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    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
    To: Bob <sips:bob@biloxi.example.com>;tag=23431
    Call-ID: 12345600@atlanta.example.com
    CSeq: 1 INVITE
    Contact: <sips:bob@biloxi.example.com>



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




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    /* 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>
    Call-ID: 4802029847@biloxi.example.com
    CSeq: 1 INVITE
    Contact: <sips:bob@biloxi.example.com>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY



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    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>
    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
    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



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    /* 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. */

    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
    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



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    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
    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



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    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 -> 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 Source <sips:music@source.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.  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 will be when the remote UA desires to
   put 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.



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   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 a
   Replaces headers that replaces 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[8] is supported, the MOH
   source may need to send requests as part of ICE negotiation[9] 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.

   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.  Payload Type Numbers

   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 [2] (section 6.1, at the very end) specifies that the payload
   type numbers used in either direction of RTP are the ones specified



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   in the SDP sent by the recipient of the RTP, which in this case is
   the remote UA, which composed the offer.

   But strict compliance to [2] (section 8.3.2) requires that payload
   type numbers used in the SDP answer may duplicate the payload type
   numbers used in any offers and answers previously used in the dialog
   only 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 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 [2].

   We can prevent this problem by utilizing the requirement itself to
   control the behavior of the MOH source: When the executing UA is
   composing the INVITE to the MOH source, it compiles a list of all the
   (dynamically-assigned) payload type numbers which have been used in
   the original dialog but which are not mapped to a media format in the
   offer SDP.  (The executing UA must be maintaining a list of all
   previously used payload type numbers anyway, in order to comply with
   [2].)  Then, for each of these payload type numbers, it inserts
   session-level or media-level (as appropriate) a=rtpmap lines
   specifying the payload type number and the media format that it has
   been used for.  Because of the reuse rule, the MOH source cannot
   propose those payload type numbers for any other media format.

   However, once the payload type numbers have been defined within the
   MOH dialog, the executing UA need not insert additional a=rtpmap
   lines in later SDP that is passed through.




















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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 are sent directly from the music-on-hold
       source to the remote UA, rather than being relayed through the
       executing UA.

   3.  The remote UA sees, in the incoming SDP, the address/port that
       the MOH source 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]).

   5.  It strictly complies with the payload type number rules.[2]























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4.  Security Considerations

   Some UAs filter incoming media based on the address of origin in
   order to avoid SPIT.  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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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 the sequence of operations, allowing improvements to
   the SDP handling, were suggested by Venkatesh[7].

   John Elwell[9] pointed out the need for the executing UA to pass
   through re-INVITEs/UPDATEs in order to allow ICE negotiation.

   Paul Kyzivat[10] pointed out the difficulties regarding re-use of
   payload type numbers.







































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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.

6.2.  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.






























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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]   Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
         Protocol for Network Address Translator (NAT) Traversal for
         Offer/Answer Protocols", I-D draft-ietf-mmusic-ice-19,
         October 2007.

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

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

   [11]  Sawada, T. and P. Kyzivat, "SIP (Session Initiation Protocol)
         Usage of the Offer/Answer Model",
         I-D draft-ietf-sipping-sip-offeranswer-04, October 2007.



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Author's Address

   Dale R. Worley
   Nortel Networks Corp.
   600 Technology Park Dr.
   Billerica, MA  01821
   US

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









































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Full Copyright Statement

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