Network Working Group B. Campbell

Network Working Group B. Campbell
Request for Comments: 3087 R. Sparks
Category: Informational dynamicsoft
April 2001

Control of Service Context using SIP Request-URI

Status of this Memo

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

Copyright Notice

Copyright (C) The Internet Society (2001). All Rights Reserved.

Abstract

This memo provides information for the Internet community. It
describes a useful way to conceptualize the use of the standard SIP
(Session Initiation Protocol) Request-URI (Uniform Resource
Identifier) that the authors and many members of the SIP community
think is suitable as a convention. It does not define any new
protocol with respect to RFC 2543.

In a conventional telephony environment, extended service
applications often use call state information, such as calling party,
called party, reason for forward, etc, to infer application context.
In a SIP/2.0 call, much of this information may be either non-
existent or unreliable. This document proposes a mechanism to
communicate context information to an application. Under this
proposal, a client or proxy can communicate context through the use
of a distinctive Request-URI. This document continues with examples
of how this mechanism could be used in a voice mail application.

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Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 3
2. Example Application . . . . . . . . . . . . . . . . . . . 3
2.1 Using URIs to Control Voice Mail Service Behavior . . . . 3
3. Voice Mail Scenario Descriptions . . . . . . . . . . . . . 5
3.1 Deposits . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1.1 Direct Request to Deposit to a particular mailbox . . . . 5
3.1.1.1 SIP source . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1.1.2 Arbitrary PSTN source . . . . . . . . . . . . . . . . . . 5
3.1.1.3 Recognized PSTN source . . . . . . . . . . . . . . . . . . 6
3.1.2 Direct Request to Deposit, mailbox to be determined . . . 6
3.1.2.1 SIP source . . . . . . . . . . . . . . . . . . . . . . . . 6
3.1.2.2 PSTN source . . . . . . . . . . . . . . . . . . . . . . . 6
3.1.2.3 Indirect Request to Deposit, due to find-me proxy decision 6
3.2 Retrievals . . . . . . . . . . . . . . . . . . . . . . . . 7
3.2.1 Request to Retrieve from a particular mailbox . . . . . . 7
3.2.1.1 Trusted SIP source . . . . . . . . . . . . . . . . . . . . 7
3.2.1.2 Authenticated SIP source . . . . . . . . . . . . . . . . . 7
3.2.1.3 Unauthenticated SIP source . . . . . . . . . . . . . . . . 7
3.2.1.4 PSTN source . . . . . . . . . . . . . . . . . . . . . . . 7
3.2.2 Request to Retrieve, mailbox to be determined . . . . . . 7
3.2.2.1 SIP source . . . . . . . . . . . . . . . . . . . . . . . . 7
3.2.2.2 Arbitrary PSTN source . . . . . . . . . . . . . . . . . . 8
3.2.2.3 Recognized PSTN source . . . . . . . . . . . . . . . . . . 8
4. Voice Mail Call Flow Examples . . . . . . . . . . . . . . 8
4.1 Generic Scenario . . . . . . . . . . . . . . . . . . . . . 8
4.1.1 Direct call to the voice mail system . . . . . . . . . . . 8
4.2 Message Deposit Scenarios . . . . . . . . . . . . . . . . 13
4.2.1 Call to known subscriber forwarded on no answer . . . . . 13
4.2.2 Call to known subscriber forwarded on busy . . . . . . . . 19
4.2.3 Direct call to a subscriber's mailbox . . . . . . . . . . 24
4.3 Message Retrieval Scenarios . . . . . . . . . . . . . . . 29
4.3.1 Call to retrieve messages believed to be from a known
subscriber . . . . . . . . . . . . . . . . . . . . . . . . 29
4.3.2 Call to retrieve messages from an authenticated subscriber 33
5. Security Considerations . . . . . . . . . . . . . . . . . 38
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . 38
References . . . . . . . . . . . . . . . . . . . . . . . . 38
Authors' Addresses . . . . . . . . . . . . . . . . . . . . 38
Full Copyright Statement . . . . . . . . . . . . . . . . . 39

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1. Introduction

A communication service should make use of the information it has at
hand when being accessed. For example, in most current voice mail
implementations, a subscriber retrieving messages from his own desk
does not have to reenter his voice mailbox number - the service
assumes that the store being accessed is the one associated with the
endpoint being used to access the service. Some services allow the
user to validate this assumption using IVR techniques before
prompting for a PIN.

This concept of context-awareness can be captured in a voice mail
service implementing SIP as defined in RFC 2543[1], without
modification, through the standard use of that protocol's Request-
URI. Furthermore, the concept is applicable to any SIP-based service
where initial application state should be determined from context.

This concept is a usage convention of standard SIP as defined in RFC
2543[1] and does not modify or extend that protocol in any way.

2. Example Application

In this document, we use the example of voice mail to illustrate the
technique. One motivation for applying this technique to this
problem is allowing a proxy or location server to control the initial
state of a voice service. For example, a voice client might register
a contact list ending with the URL that would accept voice messages
for the client.

2.1 Using URIs to Control Voice Mail Service Behavior

Many conventional voice mail systems use call state information, such
as the calling party, called party, reason for forward, etc, to
decide the initial application state. For example, it might play one
outgoing message if the call reached voice mail because the called
party did not answer and another if the line was busy. It decides
whom the message is for based on the called party information. If
the call originated from a subscriber's phone number, it might
authenticate the caller and then go directly to the message retrieval
and account maintenance menu.

When a new subscriber is added to a system, a set of identities could
be generated, each given a unique sip URI. The following tables show
some of the identities that might be generated (it is not
exhaustive). The example schemes show that the URIs could, but don't
necessarily have to, have mnemonic value.

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In practical applications, it is important that an application does
not apply semantic rules to the various URIs. Instead, it should
allow any arbitrary string to be provisioned, and map the string to
the desired behavior. The owner of the system may choose to
provision mnemonic strings, but the application should not require
it. In any large installation, the system owner is likely to have
pre-existing rules for mnemonic URIs, and any attempt by an
application to define its own rules may create a conflict. For our
example, this means a voice mail system should allow an arbitrary mix
of URLs from these schemes, or any other scheme that renders valid
SIP URIs to be provisioned, rather than enforce one particular
scheme.

URI Identity Example Scheme 1
Example Scheme 2
Example Scheme 3

Deposit with sip:[email protected]
standard greeting sip:[email protected]
sip:[email protected];mode=deposit

Deposit with on sip:sub-rjs-deposit-busy.vm.wcom.com
phone greeting sip:[email protected]
sip:[email protected];mode=3991243

Deposit with sip:[email protected]
special greeting sip:[email protected]
sip:[email protected];mode=sg

Retrieve - SIP sip:[email protected]
authentication sip:[email protected]
sip:[email protected];mode=retrieve

Retrieve - prompt sip:sub-rjs-retrieve-inpin.vm.wcom.com
for PIN in-band sip:[email protected]
sip:[email protected];mode=inpin

When a service is first set up, identities such as the following
could be created.

URI Identity Example Scheme 1
Example Scheme 2
Example Scheme 3

Deposit - sip:[email protected]
identify target sip:[email protected]
mailbox by To: sip:[email protected]

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RFC 3087 SIP Service Control April 2001

Retrieve - sip:[email protected]
identify target sip:[email protected]
mailbox by SIP sip:[email protected]
authentication

Deposit - prompt sip:[email protected]
for target sip:[email protected]
mailbox in-band sip:[email protected];mode=inband

Retrieve - prompt sip:[email protected]
for target sip:[email protected]
mailbox and PIN sip:[email protected];mode=inband
in-band

In addition to providing this set of URIs to the subscriber (to use
as he sees fit), an integrated service provider could add these to
the set of contacts in a find-me proxy. The proxy could then route
calls to the appropriate URI based on the origin of the request, the
subscriber's preferences and current state.

3. Voice Mail Scenario Descriptions

In each of these scenarios, the PSTN gateway is configured to
communicate only with a particular proxy-registrar.

3.1 Deposits

3.1.1 Direct Request to Deposit to a particular mailbox

3.1.1.1 SIP source

A SIP client that knew the URI for a particular deposit mailbox
(sip:[email protected]) could place a direct invitation to
the voicemail service, or through a protecting proxy. The proxy
could restrict access to deposit identities with special greetings by
authenticating the requester.

3.1.1.2 Arbitrary PSTN source

The gateway's proxy would map a call from an unrecognized PSTN number
to a number associated with a subscriber's mailbox into an invite to
the deposit with standard greeting URI (sip:sub-rjs-
[email protected]).

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3.1.1.3 Recognized PSTN source

The gateway's proxy would map a call from a recognized (exact or
pattern match) PSTN number to a number associated with a subscriber's
mailbox into an invite to the appropriate special greeting URI
(sip:[email protected]). The gateway's ability to
identify the calling party (using calling party number) is trusted,
so no further authentication of the requester is performed.

3.1.2 Direct Request to Deposit, mailbox to be determined

3.1.2.1 SIP source

A voice mail service or its protecting proxy could expose a generic
deposit URL for use when a caller wished to go directly to voice
mail. The service would likely play an IVR dialog to determine what
message store to deposit a message into.

An application designer may be tempted to attempt to match the To:
and From: headers on a call to infer information. However, this
approach could cause complications when multiple proxy forwards occur
in a call. For example, A calls B, who has all calls forwarded to C.
C forwards the call to her voice mail service. If the voice mail
service matches the To: header to determine the message store, it
will get the information for B instead of C. But there is no reason
to assume that C's voice mail service has any knowledge of B.

3.1.2.2 PSTN source

The gateway's proxy would map a call from an unrecognized PSTN number
to the top level voice mail service access number to an invite to the
Deposit - prompt for target mailbox in-band URI (sip:deposit-
[email protected] for example). Getting the call to the target mailbox
would proceed as in the SIP source case.

3.1.2.3 Indirect Request to Deposit, due to find-me proxy decision

A find-me proxy could map an invitation to a subscriber
(sip:[email protected]) to the appropriate voice mail service URI
depending on the subscriber's current state. The normal deposit URI
could be chosen if the subscriber's contact list has been otherwise
exhausted with no answer. The busy-announcement URI would be chosen
when a busy everywhere response is received from one of the contacts.
A DND announcement URI could be selected if the subscriber had
activated DND. Calls to sip:[email protected] could be configured
to roll to sip:[email protected]

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3.2 Retrievals

3.2.1 Request to Retrieve from a particular mailbox

3.2.1.1 Trusted SIP source

A request to retrieve the contents of a particular mailbox (sip:sub-
[email protected]) coming from a trusted source could be
honored without further authentication checks. A trusted source is
one with which the voice mail service has secure communications, and
to which it is willing to delegate authentication. This could be the
service's protecting proxy for example.

3.2.1.2 Authenticated SIP source

A service, or its protecting proxy, could choose to honor a retrieve
request for a particular mailbox (sip:[email protected])
based on SIP authentication. If SIP level authentication failed, the
service or proxy could be configured to send the call to the in-band
pin prompting URI (sip:[email protected]).

3.2.1.3 Unauthenticated SIP source

A service, or its protecting proxy, receiving a retrieve request for
a particular mailbox (sip:[email protected]) with no other
method of authenticating the requestor could send the request to the
in-band pin prompting URI (sip:[email protected]).

3.2.1.4 PSTN source

This scenario assumes that the service provider's network has been
configured such that a PSTN number could be dialed explicitly for
retrieving messages from a particular mailbox. Such services
currently exist, but are not common. In such a network, the
gateway's proxy would map the call to the in-band pin prompting URI
(sip:[email protected]).

3.2.2 Request to Retrieve, mailbox to be determined

3.2.2.1 SIP source

As in the Request to Deposit scenario, when a service receives a
request for the top level retrieve URI it would most likely need to
use in-band IVR techniques to determine the target mailbox and
authenticate the caller.

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3.2.2.2 Arbitrary PSTN source

This scenario assumes there is a single PSTN number that subscribers
dial to access the voice mail service to retrieve messages. This is
the most common access method provided by current voice mail
services.

The gateway's proxy would map a call to the top level PSTN number to
the top level retrieve in-band prompting URI (sip:retrieve-
[email protected]). Once the system identifies the target mailbox, the
call would be transferred to the appropriate in-band pin prompting
URI (sip:[email protected]).

3.2.2.3 Recognized PSTN source

This scenario also assumes there is a single PSTN number that
subscribers dial to access the voice mail service to retrieve
messages.

The gateway's proxy would recognize the calling party number as a
subscriber, and map the call to the subscriber's in-band prompting
URI (sip:[email protected])

4. Voice Mail Call Flow Examples

The following section describes some example call flows for a
hypothetical voice mail service, with the host name of vm.wcom.com.
All the call flows assume that a proxy protects the voice mail
service and that a trust relationship exists between the voice mail
service and the proxy.

4.1 Generic Scenario

4.1.1 Direct call to the voice mail system

User A calls the voice mail system directly. The voice mail system
invokes the top-level menu, which might prompt the caller for an
extension or the first few letters of a name.

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RFC 3087 SIP Service Control April 2001

User A Proxy VM Service
| | |
| INVITE F1 | |
|------------------>| |
| | INVITE F2 |
| (100 Trying) F3 |---------------------->|
|<------------------| |
| | 180 Ringing F4 |
| 180 Ringing F5 |<----------------------|
|<------------------| |
| | 200 OK F6 |
| 200 OK F6 |<----------------------|
|<------------------| |
| | |
| ACK F8 | |
|------------------>| ACK F9 |
| |---------------------->|
| | |
| RTP Established- Play top level menu |
|<-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m->|
| | |
| BYE F10 | |
|------------------>| BYE F11 |
| |---------------------->|
| | |
| | 200 OK F12 |
| |<----------------------|
| 200 OK F13 | |
|<------------------| |
| | |

Flow Id Comments

INVITE F1 INVITE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cc4e341ae6cbe5a359", opaque="",
uri="sip:[email protected]", response=<appropriately
calculated hash goes here>
Content-Type: application/sdp
Content-Length: <appropriate value>

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RFC 3087 SIP Service Control April 2001

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

/*Client for A prepares to receive data on port 49170
from the network. */

INVITE F2 INVITE sip:[email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

(100 Trying SIP/2.0 100 Trying
F3 Via: SIP/2.0/UDP here.com:5060
Proxy->A) From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

180 Ringing SIP/2.0 180 Ringing
F4 Via: SIP/2.0/UDP wcom.com:5060; branch=1
VM->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

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180 Ringing SIP/2.0 180 Ringing
F5 Via: SIP/2.0/UDP here.com:5060
Proxy->A From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

200 OK F6 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: VoiceMailSystem <sip:[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

200 OK F7 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Contact VoiceMailSystem <sip:[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

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ACK F8 ACK sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

ACK F9 ACK sip:[email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>; tag=3145678
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

/* RTP streams are established between A and VM. VM
system starts IVR dialog for top level menu */

/* User A Hangs Up with VM system. Alternatively, the
VM system could initiate the BYE*/

BYE F10 BYE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip: [email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

BYE F11 BYE sip: [email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

200 OK F12 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]

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CSeq: 2 BYE
Content-Length: 0

200 OK F13 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

4.2 Message Deposit Scenarios

4.2.1 Call to known subscriber forwarded on no answer

User A attempts to call UserB, who does not answer. The call is
forwarded to UserB's mailbox, and the voice mail system plays UserB's
outgoing message for a ring-no-answer. The flow assumes that the URL
of "sip:[email protected] maps" to the desired behavior for
depositing a message on a forward-no-answer.

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RFC 3087 SIP Service Control April 2001

User A Proxy User B VM System
| | | |
| INVITE F1 | | |
|---------------->| INVITE F2 | |
| |----------------->| |
| (100 Trying) F3 | | |
|<----------------| 180 Ringing F4 | |
| |<-----------------| |
| 180 Ringing F5 | | |
|<----------------| (Request Timeout)| |
| | | |
| | Cancel F6 | |
| |----------------->| |
| | | |
| | 200 OK F7 | |
| |<-----------------| |
| | | |
| | INVITE F8 |
| |---------------------------------->|
| | | |
| | 200 OK F9| |
| 200 OK F10 |<----------------------------------|
|<----------------| | |
| | | |
| ACK F11 | | |
|---------------->| ACK F12 | |
| |---------------------------------->|
| | | |
| RTP Established Both Ways-Deposit Msg for B |
|<-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m->|
| | | |
| BYE F13 | | |
|---------------->| BYE F14 | |
| |---------------------------------->|
| | | |
| | OK F15 | |
| OK F16 |<----------------------------------|
|<----------------| | |
| | | |

Flow Id Comments

INVITE F1 INVITE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>
Call-Id: [email protected]

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RFC 3087 SIP Service Control April 2001

CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:[email protected]", response=<appropriately
calculated hash goes here>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

/*Client for A prepares to receive data on port 49170
from the network. */

INVITE F2 INVITE sip:[email protected] SIP/2.0
Proxy->B1 Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

(100 Trying SIP/2.0 100 Trying
F3 Via: SIP/2.0/UDP here.com:5060
Proxy->A) From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

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180 Ringing SIP/2.0 180 Ringing
F4 Via: SIP/2.0/UDP wcom.com:5060; branch=1
B1->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

180 Ringing SIP/2.0 180 Ringing
F5 Via: SIP/2.0/UDP here.com:5060
Proxy->A From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

/* B1 rings for 9 seconds, this duration is a
configurable parameter in the Proxy Server. Proxy
sends Cancel and proceeds down the list of routes,
eventually hitting the voice mail URI for forward no
answer */

CANCEL F6 CANCEL sip:[email protected] SIP/2.0
Proxy->B1 Via: SIP/2.0/UDP wcom.com:5060; branch=1
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

200 OK F7 SIP/2.0 200 OK
B1->Proxy Via: SIP/2.0/UDP wcom.com:5060; branch=1
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 CANCEL
Content-Length: 0

INVITE F8 INVITE sip:[email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060;branch=2
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE

Campbell & Sparks Informational [Page 16]

RFC 3087 SIP Service Control April 2001

Contact: TheBigGuy <sip:[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

200 OK F9 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060; branch=2
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=123456
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheLittleGuyVoiceMail <sip:UserB-dep-
[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

200 OK F10 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=123456
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheLittleGuyVoiceMail <sip:UserB-dep-
[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com

Campbell & Sparks Informational [Page 17]

RFC 3087 SIP Service Control April 2001

s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

ACK F11 ACK sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip: [email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=123456
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

ACK F12 ACK sip:[email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=123456
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

/* RTP streams are established between A and B2. VM
system starts IVR dialog for message-deposit on no-
answer for UserB */

/* User A Hangs Up with VM system. Alternatively, the
VM system could initiate the BYE*/

BYE F13 BYE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip: [email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=123456
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

BYE F14 BYE sip: [email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=123456
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

Campbell & Sparks Informational [Page 18]

RFC 3087 SIP Service Control April 2001

200 OK F15 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=123456
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

200 OK F16 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=123456
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

4.2.2 Call to known subscriber forwarded on busy

User A attempts to call UserB, who is busy. The call is forwarded to
UserB's mailbox, and the voice mail system plays UserB's outgoing
message for a busy. This flow assumes that "sip:UserB-dep-
[email protected]" maps to UserB's mailbox and the behavior of "deposit
message on busy."

Campbell & Sparks Informational [Page 19]

RFC 3087 SIP Service Control April 2001

User A Proxy User B VM System
| | | |
| INVITE F1 | | |
|---------------->| INVITE F2 | |
| |----------------->| |
| (100 Trying) F3 | | |
|<----------------| 486 Busy Here F4 | |
| |<-----------------| |
| | | |
| | ACK F5 | |
| |----------------->| |
| | | |
| | INVITE F6 |
| |---------------------------------->|
| | | |
| | 200 OK F7| |
| 200 OK F8 |<----------------------------------|
|<----------------| | |
| | | |
| ACK F9 | | |
|---------------->| ACK F10 | |
| |---------------------------------->|
| | | |
| RTP Established Both Ways-Deposit Msg for B |
|<-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m->|
| | | |
| BYE F11 | | |
|---------------->| BYE F12 | |
| |---------------------------------->|
| | | |
| | OK F13 | |
| OK F14 |<----------------------------------|
|<----------------| | |
| | | |

Flow Id Comments

INVITE F1 INVITE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:[email protected]", response=<appropriately

Campbell & Sparks Informational [Page 20]

RFC 3087 SIP Service Control April 2001

calculated hash goes here>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

/*Client for A prepares to receive data on port 49170
from the network. */

INVITE F2 INVITE sip:[email protected] SIP/2.0
Proxy->B1 Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

(100 Trying SIP/2.0 100 Trying
F3 Via: SIP/2.0/UDP here.com:5060
Proxy->A) From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

486 Busy SIP/2.0 486 Busy Here
Here F4 Via: SIP/2.0/UDP wcom.com:5060;branch=1
B1->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=123456

Campbell & Sparks Informational [Page 21]

RFC 3087 SIP Service Control April 2001

Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

ACK F5 ACK sip: [email protected] SIP/2.0
Proxy->B Via: SIP/2.0/UDP wcom.com:5060; branch=1
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=123456
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

INVITE F6 INVITE sip:[email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060;branch=2
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

200 OK F7 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060; branch=2
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheLittleGuyVoiceMail <sip:UserB-dep-
[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP

Campbell & Sparks Informational [Page 22]

RFC 3087 SIP Service Control April 2001

c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

200 OK F8 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Contact TheLittleGuyVoiceMail <sip:UserB-dep-
[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

ACK F9 ACK sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

ACK F10 ACK sip:[email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

/* RTP streams are established between A and B2. VM
system starts IVR dialog for message-deposit on busy
for UserB */

Campbell & Sparks Informational [Page 23]

RFC 3087 SIP Service Control April 2001

/* User A Hangs Up with VM system. Alternatively, the
VM system could initiate the BYE*/

BYE F11 BYE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

BYE F12 BYE sip: [email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

200 OK F13 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

200 OK F14 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuy <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

4.2.3 Direct call to a subscriber's mailbox

User A calls UserB's mailbox directly. This flow assumes that
"sip:[email protected]" maps to UserB's mailbox and the behavior
of "generic message deposit"

Campbell & Sparks Informational [Page 24]

RFC 3087 SIP Service Control April 2001

User A Proxy VM Service
| | |
| INVITE F1 | |
|------------------>| |
| | INVITE F2 |
| (100 Trying) F3 |---------------------->|
|<------------------| |
| | 200 OK F4 |
| 200 OK F5 |<----------------------|
|<------------------| |
| | |
| ACK F6 | |
|------------------>| ACK F7 |
| |---------------------->|
| | |
| RTP Both Ways - Deposit Msg for B |
|<-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m->|
| | |
| BYE F8 | |
|------------------>| BYE F9 |
| |---------------------->|
| | |
| | 200 OK F10 |
| |<----------------------|
| 200 OK F11 | |
|<------------------| |
| | |

Flow Id Comments

INVITE F1 INVITE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:[email protected]", response=<appropriately
calculated hash goes here>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP

Campbell & Sparks Informational [Page 25]

RFC 3087 SIP Service Control April 2001

c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

/*Client for A prepares to receive data on port 49170
from the network. */

INVITE F2 INVITE sip:[email protected] SIP/2.0
Proxy->B1 Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

(100 Trying SIP/2.0 100 Trying
F3 Via: SIP/2.0/UDP here.com:5060
Proxy->A) From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

200 OK F4 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:UserB-
[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheLittleGuyVoiceMail <sip:UserB-
[email protected]>
Content-Type: application/sdp

Campbell & Sparks Informational [Page 26]

RFC 3087 SIP Service Control April 2001

Content-Length: <appropriate value>
v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

200 OK F5 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:UserB-
[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Contact TheLittleGuyVoiceMail <sip:UserB-
[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

ACK F6 ACK sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:UserB-
[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

ACK F7 ACK sip:[email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:UserB-
[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 ACK

Campbell & Sparks Informational [Page 27]

RFC 3087 SIP Service Control April 2001

Content-Length: 0
/* RTP streams are established between A and VM. VM
system starts IVR dialog for generic message-deposit
for UserB */

/* User A Hangs Up with VM system. Alternatively, the
VM system could initiate the BYE*/

BYE F8 BYE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:UserB-
[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

BYE F9 BYE sip: [email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:UserB-
[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

200 OK F10 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:UserB-
[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

200 OK F11 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: TheLittleGuyVoiceMail <sip:UserB-
[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

Campbell & Sparks Informational [Page 28]

RFC 3087 SIP Service Control April 2001

4.3 Message Retrieval Scenarios

4.3.1 Call to retrieve messages believed to be from a known subscriber

Some user uses a SIP client on UserA's desk to call the voice mail
system to retrieve messages. The SIP client has authenticated itself
to the proxy using credentials assigned to the device. The proxy can
make a weak assumption that the caller is the device owner. The URI
of "sip:[email protected]" maps to UserA's mailbox and the
behavior of "retrieve messages after prompting for and verifying
PIN." The VM System trusts the proxy, and will not accept calls from
an untrusted source. The proxy will not allow direct calls to
[email protected]. The proxy will forward calls placed to
[email protected] to [email protected] only for calls
placed from a client device assigned to UserA.

User A Proxy VM Service
| | |
| INVITE F1 | |
|------------------>| |
| | INVITE F2 |
| (100 Trying) F3 |---------------------->|
|<------------------| |
| | 200 OK F4 |
| 200 OK F5 |<----------------------|
|<------------------| |
| | |
| ACK F6 | |
|------------------>| ACK F7 |
| |---------------------->|
| | |
| RTP Both Ways - VM prompts for PIN
|<-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m->|
| | |
| BYE F8 | |
|------------------>| BYE F9 |
| |---------------------->|
| | |
| | 200 OK F10 |
| |<----------------------|
| 200 OK F11 | |
|<------------------| |
| | |

Campbell & Sparks Informational [Page 29]

RFC 3087 SIP Service Control April 2001

Flow Id Comments

INVITE F1 INVITE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Proxy-Authorization:Digest username="UserAPhone",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:[email protected]", response=<appropriately
calculated hash goes here>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

/*Client for A prepares to receive data on port 49170
from the network. */

INVITE F2 INVITE sip:[email protected] SIP/2.0
Proxy->B1 Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

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RFC 3087 SIP Service Control April 2001

(100 Trying SIP/2.0 100 Trying
F3 Via: SIP/2.0/UDP here.com:5060
Proxy->A) From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

200 OK F4 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: VoiceMailSystem <sip:UserA-
[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

200 OK F5 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Contact VoiceMailSystem <sip: UserA-
[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

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RFC 3087 SIP Service Control April 2001

ACK F6 ACK sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

ACK F7 ACK sip:[email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>; tag=3145678
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

/* RTP streams are established between A and VM. VM
determines that the call is likely from UserA, and
starts a message retrieval session, prompting for
PIN*/

/* User A Hangs Up with VM system. Alternatively, the
VM system could initiate the BYE*/

BYE F8 BYE sip: [email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

BYE F9 BYE sip: [email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

200 OK F10 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>

Campbell & Sparks Informational [Page 32]

RFC 3087 SIP Service Control April 2001

To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

200 OK F11 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

4.3.2 Call to retrieve messages from an authenticated subscriber

UserA to call the voice mail system to retrieve messages.
Assumptions: The caller is authenticated using UserA's credentials.
"sip:[email protected]" maps to UserA's mailbox and the
behavior of "retrieve messages." The voice mail service trusts the
proxy not to forward any calls to that URI unless the call is
authenticated to be from UserA.

Given these assumptions, The VM service may choose not require a PIN
for calls to this URI.

Campbell & Sparks Informational [Page 33]

RFC 3087 SIP Service Control April 2001

User A Proxy VM Service
| | |
| INVITE F1 | |
|------------------>| |
| | INVITE F2 |
| (100 Trying) F3 |---------------------->|
|<------------------| |
| | 200 OK F4 |
| 200 OK F5 |<----------------------|
|<------------------| |
| | |
| ACK F6 | |
|------------------>| ACK F7 |
| |---------------------->|
| | |
| RTP Both Ways - Deposit Msg for B |
|<-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m->|
| | |
| BYE F8 | |
|------------------>| BYE F9 |
| |---------------------->|
| | |
| | 200 OK F10 |
| |<----------------------|
| 200 OK F11 | |
|<------------------| |
| | |

Flow Id Comments

INVITE F1 INVITE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:[email protected]", response=<appropriately
calculated hash goes here>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP

Campbell & Sparks Informational [Page 34]

RFC 3087 SIP Service Control April 2001

c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

/*Client for A prepares to receive data on port 49170
from the network. */

INVITE F2 INVITE sip:[email protected] SIP/2.0
Proxy->B1 Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: TheBigGuy <sip:[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000

(100 Trying SIP/2.0 100 Trying
F3 Via: SIP/2.0/UDP here.com:5060
Proxy->A) From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>
Call-Id: [email protected]
CSeq: 1 INVITE
Content-Length: 0

200 OK F4 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: VoiceMailSystem <sip:UserA-retrieve-
[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

Campbell & Sparks Informational [Page 35]

RFC 3087 SIP Service Control April 2001

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

200 OK F5 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 INVITE
Contact VoiceMailSystem <sip: UserA-retrieve-
[email protected]>
Content-Type: application/sdp
Content-Length: <appropriate value>

v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000

ACK F6 ACK sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route: <sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

ACK F7 ACK sip:[email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>; tag=3145678
Call-Id: [email protected]
CSeq: 1 ACK
Content-Length: 0

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RFC 3087 SIP Service Control April 2001

/* RTP streams are established between A and VM. VM
determines that the call is likely from UserA, and
starts a message retrieval session. Since the proxy
has already authenticated the identity of UserA, the
VM does not need to prompt for PIN. */

/* User A Hangs Up with VM system. Alternatively, the
VM system could initiate the BYE*/

BYE F8 BYE sip:[email protected] SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:[email protected]>
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

BYE F9 BYE sip: [email protected] SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

200 OK F10 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

200 OK F11 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:[email protected]>
To: VoiceMail <sip:[email protected]>;tag=3145678
Call-Id: [email protected]
CSeq: 2 BYE
Content-Length: 0

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RFC 3087 SIP Service Control April 2001

5. Security Considerations

This document discusses a usage of SIP/2.0 as defined by RFC 2543[1].
It introduces no additions, modifications, or restrictions to the
protocol defined therein. Any implementation of the concepts in this
document is subject to the issues discussed there.

6. Acknowledgments

The authors would like to thank Chris Cunningham, Steve Donovan, Alan
Johnston, Henry Sinnreich, Kevin Summers, John Truetken, and Dean
Willis for their discussion of and contribution to this work.

References

[1] Handley, M., Schulzrinne, H., Schooler, E. and J. Rosenberg,
"SIP: Session Initiation Protocol", RFC 2543, March 1999.

Authors' Addresses

Ben Campbell
dynamicsoft
5100 Tennyson Parkway
Suite 1200
Plano, TX 75024

EMail: [email protected]

Robert J. Sparks
dynamicsoft
5100 Tennyson Parkway
Suite 1200
Plano, TX 75024

EMail: [email protected]

Campbell & Sparks Informational [Page 38]

RFC 3087 SIP Service Control April 2001

Full Copyright Statement

Copyright (C) The Internet Society (2001). All Rights Reserved.

This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.

The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.

This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Acknowledgement

Funding for the RFC Editor function is currently provided by the
Internet Society.

Campbell & Sparks Informational [Page 39]