Internet Engineering Task Force B. Campbell
Internet-Draft R. Sparks
Expires: July 2, 2001 dynamicsoft
January 2001
Control of Service Context using SIP Request-URI
draft-campbell-sip-service-control-02
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
Internet-Drafts are working documents of the Internet Engineering
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This Internet-Draft will expire on July 2, 2001.
Copyright Notice
Copyright (C) The Internet Society (2001). All Rights Reserved.
Abstract
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 draft 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 draft 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 . . . . . . . . 18
4.2.3 Direct call to a subscriber's mailbox . . . . . . . . . . 23
4.3 Message Retrieval Scenarios . . . . . . . . . . . . . . . 28
4.3.1 Call to retrieve messages believed to be from a known
subscriber . . . . . . . . . . . . . . . . . . . . . . . . 28
4.3.2 Call to retrieve messages from an authenticated subscriber 32
5. Security Considerations . . . . . . . . . . . . . . . . . 36
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . 36
References . . . . . . . . . . . . . . . . . . . . . . . . 36
Authors' Addresses . . . . . . . . . . . . . . . . . . . . 36
Full Copyright Statement . . . . . . . . . . . . . . . . . 37
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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.
2. Example Application
In this draft, 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.
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
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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:sub-rjs-deposit@vm.wcom.com
standard greeting sip:677283@vm.wcom.com
sip:rjs@vm.wcom.com;mode=deposit
Deposit with on sip:sub-rjs-deposit-busy.vm.wcom.com
phone greeting sip:677372@vm.wcom.com
sip:rjs@vm.wcom.com;mode=3991243
Deposit with sip:sub-rjs-deposit-sg@vm.wcom.com
special greeting sip:677384@vm.wcom.com
sip:rjs@vm.wcom.com;mode=sg
Retrieve - SIP sip:sub-rjs-retrieve@vm.wcom.com
authentication sip:677405@vm.wcom.com
sip:rjs@vm.wcom.com;mode=retrieve
Retrieve - prompt sip:sub-rjs-retrieve-inpin.vm.wcom.com
for PIN in-band sip:677415@vm.wcom.com
sip:rjs@vm.wcom.com;mode=inpin
When a service is first set up, identities such as the following
could be created.
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URI Identity Example Scheme 1
Example Scheme 2
Example Scheme 3
Deposit - sip:deposit@vm.wcom.com
identify target sip:670001@vm.wcom.com
mailbox by To: sip:deposit@vm.wcom.com
Retrieve - sip:retrieve@vm.wcom.com
identify target sip:670002@vm.wcom.com
mailbox by SIP sip:retrieve@vm.wcom.com
authentication
Deposit - prompt sip:deposit-in@vm.wcom.com
for target sip:670003@vm.wcom.com
mailbox in-band sip:deposit@vm.wcom.com;mode=inband
Retrieve - prompt sip:retrieve-in@vm.wcom.com
for target sip:670004@vm.wcom.com
mailbox and PIN sip:retrieve@vm.wcom.com;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:sub-rjs-deposit@vm.wcom.com) 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
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invite to the deposit with standard greeting URI
(sip:sub-rjs-deposit@vm.wcom.com).
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:sub-rjs-deposit-sg@vm.wcom.com). 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-in@vm.wcom.com 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:rjs@wcom.com) 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
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had activated DND. Calls to sip:receptionist@wcom.com could be
configured to roll to sip:deposit@wcom.com
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-rjs-retrieve@vm.wcom.com) 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:sub-rjs-retrieve@vm.wcom.com)
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:sub-rjs-retrieve-inpin@vm.wcom.com).
3.2.1.3 Unauthenticated SIP source
A service, or its protecting proxy, receiving a retrieve request for
a particular mailbox (sip:sub-rjs-retrieve@vm.wcom.com) with no
other method of authenticating the requestor could send the request
to the in-band pin prompting URI
(sip:sub-rjs-retrieve-inpin@vm.wcom.com).
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:sub-rjs-retrieve-inpin@vm.wcom.com).
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
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authenticate the caller.
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-in@vm.wcom.com). Once the system identifies the target
mailbox, the call would be transferred to the appropriate in-band
pin prompting URI (sip:sub-rjs-retrieve-inpin@vm.wcom.com).
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:sub-rjs-retrieve-inpin@vm.wcom.com)
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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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:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cc4e341ae6cbe5a359", opaque="",
uri="sip:VoiceMail@wcom.com", response=<appropriately
calculated hash goes here>
Content-Type: application/sdp
Content-Length: <appropriate value>
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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:top@vm.wcom.com 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:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
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:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
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:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
180 Ringing SIP/2.0 180 Ringing
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F5 Via: SIP/2.0/UDP here.com:5060
Proxy->A From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: VoiceMailSystem <sip:top@vm.wcom.com>
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:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact VoiceMailSystem <sip:top@vm.wcom.com>
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 F8 ACK sip:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
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Route:<sip:top@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
ACK F9 ACK sip:top@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>; tag=3145678
Call-Id: 12345600@here.com
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:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip: top@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
BYE F11 BYE sip: top@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
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200 OK F13 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserB-dep-fna@vm.wcom.com maps" to the desired
behavior for depositing a message on a forward-no-answer.
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 |
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|<-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:UserB@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:UserB@wcom.com", 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:UserB1@somewhere.wcom.com 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:UserB@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>
Call-Id: 12345600@here.com
Campbell & Sparks Expires July 2, 2001 [Page 14]
Internet-Draft SIP Service Control January 2001
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
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:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
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:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserB1@wcom.com SIP/2.0
Proxy->B1 Via: SIP/2.0/UDP wcom.com:5060; branch=1
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
Campbell & Sparks Expires July 2, 2001 [Page 15]
Internet-Draft SIP Service Control January 2001
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:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 CANCEL
Content-Length: 0
INVITE F8 INVITE sip:UserB-dep-fna@vm.wcom.com 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:UserB@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
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:UserB@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=123456
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheLittleGuyVoiceMail <sip:UserB-dep-
fna@vm.wcom.com>
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
Campbell & Sparks Expires July 2, 2001 [Page 16]
Internet-Draft SIP Service Control January 2001
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:UserB@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=123456
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheLittleGuyVoiceMail <sip:UserB-dep-
fna@vm.wcom.com>
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 F11 ACK sip:UserB@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip: UserB-dep-fna@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=123456
Call-Id: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
ACK F12 ACK sip:UserB-dep-fna@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=123456
Call-Id: 12345600@here.com
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*/
Campbell & Sparks Expires July 2, 2001 [Page 17]
Internet-Draft SIP Service Control January 2001
BYE F13 BYE sip:UserB@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip: UserB-dep-fna@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=123456
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
BYE F14 BYE sip: UserB-dep-fna@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=123456
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
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:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=123456
Call-Id: 12345600@here.com
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:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=123456
Call-Id: 12345600@here.com
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-fb@vm.wcom.com" maps to UserB's mailbox and the
behavior of "deposit message on busy."
Campbell & Sparks Expires July 2, 2001 [Page 18]
Internet-Draft SIP Service Control January 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:UserB@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:UserB@wcom.com", response=<appropriately
Campbell & Sparks Expires July 2, 2001 [Page 19]
Internet-Draft SIP Service Control January 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:UserB1@somewhere.wcom.com 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:UserB@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
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:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>
Call-Id: 12345600@here.com
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:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=123456
Call-Id: 12345600@here.com
Campbell & Sparks Expires July 2, 2001 [Page 20]
Internet-Draft SIP Service Control January 2001
CSeq: 1 INVITE
Content-Length: 0
ACK F5 ACK sip: UserB1@wcom.com SIP/2.0
Proxy->B Via: SIP/2.0/UDP wcom.com:5060; branch=1
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=123456
Call-Id: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
INVITE F6 INVITE sip:UserB-dep-fb@vm.wcom.com 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:UserB@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
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:UserB@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheLittleGuyVoiceMail <sip:UserB-dep-
fb@vm.wcom.com>
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
Campbell & Sparks Expires July 2, 2001 [Page 21]
Internet-Draft SIP Service Control January 2001
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:UserB@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact TheLittleGuyVoiceMail <sip:UserB-dep-
fb@vm.wcom.com>
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:UserB@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:UserB-dep-fb@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
ACK F10 ACK sip:UserB-dep-fb@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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 */
/* User A Hangs Up with VM system. Alternatively, the
VM system could initiate the BYE*/
BYE F11 BYE sip:UserB@wcom.com SIP/2.0
Campbell & Sparks Expires July 2, 2001 [Page 22]
Internet-Draft SIP Service Control January 2001
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route: <sip:UserB-dep-fnb@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
BYE F12 BYE sip: UserB-dep-fnb@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserA@here.com>
To: TheLittleGuy <sip:UserB@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserB-dep@vm.wcom.com" maps to UserB's mailbox and the behavior
of "generic message deposit"
Campbell & Sparks Expires July 2, 2001 [Page 23]
Internet-Draft SIP Service Control January 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:UserB-VM@vm.wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-VM@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:UserB-VM@wcom.com", 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 Expires July 2, 2001 [Page 24]
Internet-Draft SIP Service Control January 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:UserB-dep@vm.wcom.com 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:UserB-VM@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-VM@vm.wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
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:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-VM@wcom.com>
Call-Id: 12345600@here.com
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:UserB-VM@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-
VM@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheLittleGuyVoiceMail <sip:UserB-
dep@vm.wcom.com>
Content-Type: application/sdp
Content-Length: <appropriate value>
Campbell & Sparks Expires July 2, 2001 [Page 25]
Internet-Draft SIP Service Control January 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:UserB-VM@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-
VM@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact TheLittleGuyVoiceMail <sip:UserB-
dep@vm.wcom.com>
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:UserB-VM@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:UserB-dep@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-
VM@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
ACK F7 ACK sip:UserB-dep@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-
VM@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
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/* 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:UserB-VM@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:UserB-dep@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-
VM@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
BYE F9 BYE sip: UserB-dep@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-
VM@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-
VM@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserA@here.com>
To: TheLittleGuyVoiceMail <sip:UserB-
VM@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
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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:UserA-retrieve@vm.wcom.com" 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
UserA-retrieve@vm.wcom.com. The proxy will forward calls placed to
VoiceMail@wcom.com to UserA-retrieve@vm.wcom.com 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 | |
|<------------------| |
| | |
Flow Id Comments
INVITE F1 INVITE sip:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
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From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
Proxy-Authorization:Digest username="UserAPhone",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:VoiceMail@wcom.com", 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:UserA-retrieve@vm.wcom.com 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:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
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:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
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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:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: VoiceMailSystem <sip:UserA-
retrieve@vm.wcom.com>
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:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact VoiceMailSystem <sip: UserA-
retrieve@vm.wcom.com>
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:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:UserA-retrieve@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
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Call-Id: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
ACK F7 ACK sip:UserA-retrieve@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>; tag=3145678
Call-Id: 12345600@here.com
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: VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:UserA-retrieve@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
BYE F9 BYE sip: UserA-retrieve@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
200 OK F11 SIP/2.0 200 OK
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Proxy->A Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserA-retrieve-auth@vm.wcom.com" 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.
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
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INVITE F1 INVITE sip:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="",
uri="sip:VoiceMail@wcom.com", 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:UserA-retrieve-auth@vm.wcom.com 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:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
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:UserA@here.com>
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To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
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:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: VoiceMailSystem <sip:UserA-retrieve-
auth@vm.wcom.com>
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:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact VoiceMailSystem <sip: UserA-retrieve-
auth@vm.wcom.com>
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:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route: <sip:UserA-retrieve-auth@vm.wcom.com>
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From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
ACK F7 ACK sip:UserA-retrieve-auth@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>; tag=3145678
Call-Id: 12345600@here.com
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. 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:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:UserA-retrieve-auth@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
BYE F9 BYE sip: UserA-retrieve-auth@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 2 BYE
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Content-Length: 0
200 OK F11 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 2 BYE
Content-Length: 0
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: bcampbell@dynamicsoft.com
Robert J. Sparks
dynamicsoft
5100 Tennyson Parkway
Suite 1200
Plano, TX 75024
email: rsparks@dynamicsoft.com
Campbell & Sparks Expires July 2, 2001 [Page 36]
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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.
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