SIP D.R. Worley
Internet-Draft Bluesocket
Expires: May 10, 2008 November 7, 2007
Session Initiation Protocol Service Example -- Music on Hold
draft-worley-service-example-00
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Copyright Notice
Copyright (C) The IETF Trust (2007).
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", 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 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 . . . . . . . . . . . . . . . . . . . . . . . 4
2.3. Example Message Flow . . . . . . . . . . . . . . . . . . . 5
2.4. Alternative Example Message Flow . . . . . . . . . . . . . 13
3. Advantages . . . . . . . . . . . . . . . . . . . . . . . . . . 22
4. Security Considerations . . . . . . . . . . . . . . . . . . . 23
5. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24
5.1. Normative References . . . . . . . . . . . . . . . . . . . 24
5.2. Informative References . . . . . . . . . . . . . . . . . . 24
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 25
Intellectual Property and Copyright Statements . . . . . . . . . . 26
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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, 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 and fully standards-compliant.
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2. Technique
The essence of the technique is that when the executing UA performs a
re-INVITE of the remote UA to establish the hold state, it extracts
the answer SDP[2], 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.
2.1. Placing a Call on Hold and Providing an External Media Stream
1. The user instructs the user's UA to put the conversation on-hold.
2. The user's UA sends a re-INVITE to the remote UA with SDP that
declines to receive media. This establishes the on-hold state
(in that direction).
3. The remote UA responds 200 to the re-INVITE, and includes SDP
giving its own listening address/port, which should indicate that
it will not send media.
4. The user's UA composes and sends a new INVITE to the configured
external 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 codec numbers.
5. The MOH source responds 200 to the INVITE.
6. 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 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 user instructs the user's UA to take the conversation off-
hold.
2. The user's UA sends a re-INVITE to the remote UA with SDP that
requests to receive media.
3. When the remote UA responds 200 to the re-INVITE, the user's UA
sends BYE on 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 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:
| | |
|INVITE (hold) F5| |
|<---------------| |
| 200 OK F6 | |
|--------------->| |
| ACK F7 | |
|<---------------| |
| no RTP | |
| | |
Bob's UA initiates music-on-hold:
| | |
| | INVITE F8 |
| |------------->|
| | 200 OK F9 |
| |<-------------|
| | ACK F10 |
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| |------------->|
| Music-on-hold RTP |
|<==============================|
| | |
The music on hold is active.
Bob takes Alice off hold:
| | |
| INVITE F11 | |
|<---------------| |
| 200 OK F12 | |
|--------------->| |
| 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
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
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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>
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
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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. */
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-Type: application/sdp
Content-Length: [omitted]
v=0
o=bob 2890844527 2890844528 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
a=sendonly
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
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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 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
a=recvonly
F7 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
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Length: 0
/* Bob's UA initiates music-on-hold. */
F8 INVITE Bob -> Music Server
INVITE sips:music@server.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 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
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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
F9 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
F10 ACK Bob -> Music Server
ACK sips:music@server.example.com SIP/2.0
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
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/* 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
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 2890844528 IN IP4 atlanta.example.com
s=
c=IN IP4 atlanta.example.com
t=0 0
m=audio 49170 RTP/AVP 0
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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
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Length: 0
F14 BYE Bob -> Music Server
INVITE 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 INVITE
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 */
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2.4. Alternative Example Message Flow
A disadvantage of the previous message flow is that the RTP is sent
to the remote UA from an address that is different from that which is
given in the offer SDP of the re-INVITE. This can trigger SPIT-
prevention behavior in some UA's.[Section 4] This section shows an
alternative example message flow which avoids this problem.
Again, this is just one possible message flow that illustrates this
technique; numerous variations on these operations are allowed by the
applicable standards.
This technique involves interleaving the SDP offer/answer sequences
of the two dialogs (original call and MOH):
1. The executing UA sends the re-INVITE without SDP, forcing the
remote UA to provide an SDP offer in the 200 response.
2. The executing UA sends this offer as the SDP of the INVITE that
it sends to the music-on-hold server. (The offer is modified to
have a suitable o= line, and the media directionality is set to
"recvonly", the reverse of "on-hold".)
3. The music-on-hold server provides an SDP answer, which (we
assume) includes its media source address as its listening
address.
4. The executing UA sends this SDP answer as its SDP answer in the
ACK for the re-INVITE. (The answer is modified to have a
suitable o= line.)
Alice Bob Music Server
Alice establishes the call:
| | |
| INVITE F1 | |
|--------------->| |
| 180 Ringing F2 | |
|<---------------| |
| 200 OK F3 | |
|<---------------| |
| ACK F4 | |
|--------------->| |
| RTP | |
|<==============>| |
| | |
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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, |
| | 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 | |
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| (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
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
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;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>
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
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/* 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
/* 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
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/* 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
;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
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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
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
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. */
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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
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
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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
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Length: 0
F14 BYE Bob -> Music Server
INVITE 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 INVITE
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 */
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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 does not
change during the call.[3]
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 technique does not require complex manipulation of SDP, and
particularly does not require a SIP agent to modify received SDP
to be acceptable to be sent within an already established
sequence of SDP (which can require a complex accounting of the m=
lines).
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4. Security Considerations
SDP, by its organization, specifies what address and port a UA will
use to listen for media, but implicitly allows media to be sent to
that address and port from any address and port. Some UAs, in order
to avoid SPIT, will refuse to render media that are sent from an
address which is not the listening address for the remote UA. That
policy will also block music-on-hold that is provided using this
technique.
This problem can be circumvented in at least three ways:
The first method is to disable the media-address restriction. Since
this brings the UA's behavior into alignment with the common SIP
model, it is probably the most reliable for overall interoperability.
The second method is for the holding UA to obtain knowledge of the
media sending address of the music-on-hold server by some means, and
substitute that address for its own in the SDP in the re-INVITE that
places the dialog on-hold. This technique requires that the holding
UA can obtain this sending address, and that the technique for
specifying the "on-hold" condition in the re-INVITE is not that of
setting the listening address in the c= line to 0.0.0.0.
The third method is to use the more complex interleaved SDP offer/
answer system illustrated in Section 2.4. This technique requires
that the executing UA can coordinate the SDP offer/answer mechanism
between two dialogs.
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5. References
5.1. 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.
5.2. Informative References
[3] 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.
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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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Copyright Statement
Copyright (C) The IETF Trust (2007). This document is subject to the
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Acknowledgment
Funding for the RFC Editor function is currently provided by the
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Worley Expires May 10, 2008 [Page 26]
