Aswin,
One thing I want to clarify is , Iam not looking at the call flows.
I need details about - how the feature enabling in POTS phone will be
translated to SIP message ?.
If the feature enabling is done thro' SUBSCRIBE message , what will be event
package name for call transfer /call forward . ?
If not thro' SUBSCRIBE what will be the message used to enable a feature like
call transfer /call forward.?
Thanks,
sathish
Aswin Bhupalam <[EMAIL PROTECTED]> wrote:
Sathish,
Find attached sipping services draft. Hope this helps.
Regards,
Aswin Bhupalam.
----- Original Message -----
From: "Sathish Chandrasekaran"
To:
Sent: Monday, March 27, 2006 2:52 PM
Subject: [Sip-implementors] Query on Subscribe for Call forward feature
> All,
> I have query on "SUBSCRIBE" regarding call features like Call
forward unconditional.
> Say for example ,
> Scenario is
> - POTS phone connected to CPE( which Sends out as SIP message towards
the SIP Server )
>
> For example :
> -If the end user needs to activate call forward busy / unconditional
from POTS phone.he will be just dialing *xx followed by respective phone
number.
> 1) what would be the SIP message needs to sends out to server for
intimating server to enable the feature ?
> I feel it would be the "SUBSCRIBE" Message .If this is Ok ,
then what would be the package name ?
> Can any body give me a pointer to solve this ?
> Thanks ,
> sathish
>
>
>
>
>
> ---------------------------------
> Jiyo cricket on Yahoo! India cricket
> Yahoo! Messenger Mobile Stay in touch with your buddies all the time.
> _______________________________________________
> Sip-implementors mailing list
> [email protected]
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
SIPPING Working Group A. Johnston
Internet-Draft MCI
Expires: January 18, 2006 R. Sparks
C. Cunningham
S. Donovan
Estacado Systems
K. Summers
Sonus
July 17, 2005
Session Initiation Protocol Service Examples
draft-ietf-sipping-service-examples-09
Status of this Memo
By submitting this Internet-Draft, each author represents that any
applicable patent or other IPR claims of which he or she is aware
have been or will be disclosed, and any of which he or she becomes
aware will be disclosed, in accordance with Section 6 of BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt.
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
This Internet-Draft will expire on January 18, 2006.
Copyright Notice
Copyright (C) The Internet Society (2005).
Abstract
This document gives examples of Session Initiation Protocol (SIP)
services. This covers most features offered in so-called IP Centrex
offerings from local exchange carriers and PBX (Private Branch
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Exchange) features. Most of the services shown in this document are
implemented in the SIP User Agents, although some require the
assistance of a SIP Proxy. Some require some extensions to SIP
including the REFER, SUBSCRIBE, and NOTIFY methods and the Replaces
and Join headers. These features are not intended to be an
exhaustive set, but rather show implementations of common features
likely to be implemented on SIP IP telephones in a business
environment.
Table of Contents
1. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Service Examples . . . . . . . . . . . . . . . . . . . . . . 4
2.1 Call Hold . . . . . . . . . . . . . . . . . . . . . . . . 5
2.2 Consultation Hold . . . . . . . . . . . . . . . . . . . . 17
2.3 Music On Hold . . . . . . . . . . . . . . . . . . . . . . 34
2.4 Transfer - Unattended . . . . . . . . . . . . . . . . . . 42
2.5 Transfer - Attended . . . . . . . . . . . . . . . . . . . 49
2.6 Transfer - Instant Messaging . . . . . . . . . . . . . . . 61
2.7 Call Forwarding Unconditional . . . . . . . . . . . . . . 67
2.8 Call Forwarding - Busy . . . . . . . . . . . . . . . . . . 73
2.9 Call Forwarding - No Answer . . . . . . . . . . . . . . . 80
2.10 3-way Conference - Third Party is Added . . . . . . . . 88
2.11 3-way Conference - Third Party Joins . . . . . . . . . . 94
2.12 Single Line Extension . . . . . . . . . . . . . . . . . 99
2.13 Find-Me . . . . . . . . . . . . . . . . . . . . . . . . 117
2.14 Call Management (Incoming Call Screening) . . . . . . . 128
2.15 Call Management (Outgoing Call Screening) . . . . . . . 135
2.16 Call Park . . . . . . . . . . . . . . . . . . . . . . . 138
2.17 Call Pickup . . . . . . . . . . . . . . . . . . . . . . 148
2.18 Automatic Redial . . . . . . . . . . . . . . . . . . . . 156
2.19 Click to Dial . . . . . . . . . . . . . . . . . . . . . 161
3. Security Considerations . . . . . . . . . . . . . . . . . . 165
4. IANA Considerations . . . . . . . . . . . . . . . . . . . . 165
5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 165
6. Document History . . . . . . . . . . . . . . . . . . . . . . 166
6.1 Changes since -07 . . . . . . . . . . . . . . . . . . . . 166
7. References . . . . . . . . . . . . . . . . . . . . . . . . . 166
7.1 Normative References . . . . . . . . . . . . . . . . . . . 166
7.2 Informative References . . . . . . . . . . . . . . . . . . 167
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . 168
Intellectual Property and Copyright Statements . . . . . . . 169
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1. Overview
This document provides example call flows detailing a SIP
implementation of the following traditional telephony services:
Call Hold Music on Hold
Unattended Transfer Consultation Hold
Unconditional Call Forwarding Attended Transfer
No Answer Call Forwarding Busy Call Forwarding
Single-Line Extension 3-way Call
Incoming Call Screening Find-Me
Call Pickup Call Park
Outgoing Call Screening Automatic Redial
Click to Dial
The call flows shown in this document were developed in the design of
a SIP IP communications network. They represent an example set of
so-called IP Centrex services or PBX services.
It is the hope of the authors that this document will be useful for
SIP implementers, designers, and protocol researchers alike and will
help further the goal of a standard implementation of RFC 3261 [2]
These flows represent carefully checked and working group reviewed
scenarios of SIP service examples as a companion to the
specifications.
These call flows are based on the current version 2.0 of SIP in RFC
3261 [2] with SDP usage described in RFC 3264 [5] Other RFCs also
comprise the SIP standard and are used and references in these call
flows.
The SIP specification and the other referenced documents are
definitive as far as protocol issues are concerned. Also, these
flows do not represent the only way to implement these services -
other approaches such as 3pcc (Third Party Call Control) [17] or
Back-to-Back User Agents (B2BUA) may be more appropriate in some
circumstances. The peer-to-peer design and principles of these
service examples are described in the Multiparty Framework document
[12].
These flows assume the functionality described in the SIP Call Flow
Examples document [16], which explores basic SIP behavior. Some of
the scenarios described herein make use of the SIP method extension
REFER [3] and the SIP header extension Replaces [4], the SIP header
extension Join [9], and some of the concepts in the 3pcc (third party
call control) [17] document. The SIP Events document [6] describes
the use of SUBSCRIBE and NOTIFY while the SIP Dialog Event Package
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[8] document describes the dialog event package. Some examples make
use the GRUU (Globally Routable User Agent URI) [20] extension.
These flows were prepared assuming a network of proxies, registrars,
PSTN gateways, and other SIP servers. The use of Secure SIP URIs
(sips) is shown throughout this document with assumed certificate
validation for security. However, other security approaches such as
Digest challenges can be used.
Each call flow is presented with a textual description of the
scenario, a message flow diagram showing the messages exchanged
between separate network elements, and the detailed contents of each
message shown in the diagram.
1.1 Legend for Message Flows
Dashed lines (---) represent control messages that are mandatory to
the call scenario. These control messages can be SIP signaling.
Double dashed lines (===) represent media paths between network
elements.
Messages with parenthesis around name represent optional control
messages.
Messages are identified in the Figures as F1, F2, etc. This
references the message details in the table that follows the Figure.
Comments in the message details are shown in the following form:
/* Comments. */
2. Service Examples
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2.1 Call Hold
Alice Proxy Bob
| | |
| INVITE F1 | |
|--------------->| |
| | INVITE F2 |
|(100 Trying) F3 |------------->|
|<---------------| |
| |180 Ringing F4|
| 180 Ringing F5 |<-------------|
|<---------------| |
| | 200 OK F6 |
| 200 OK F7 |<-------------|
|<---------------| |
| ACK F8 | |
|--------------->| ACK F9 |
| |------------->|
| Both way RTP Established |
|<=============================>|
| |INVITE(hold) F10
|INVITE(hold) F11|<-------------|
|<---------------| |
| 200 OK F12 | |
|--------------->| 200 OK F13 |
| |------------->|
| | ACK F14 |
| ACK F15 |<-------------|
|<---------------| |
| No RTP Sent! |
| | INVITE F16 |
| INVITE F17 |<-------------|
|<---------------| |
| 200 OK F18 | |
|--------------->| 200 OK F19 |
| |------------->|
| | ACK F20 |
| ACK F21 |<-------------|
|<---------------| |
| Both way RTP Established |
|<=============================>|
| BYE F22 | |
|--------------->| BYE F23 |
| |------------->|
| | 200 OK F24 |
| 200 OK F25 |<-------------|
|<---------------| |
| | |
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In this scenario, Alice calls Bob, then Bob places the call on hold.
Bob then takes call off hold. Alice hangs up call. Note that hold
is unidirectional in nature. However, a UA that places the other
party on hold will generally also stop sending media, resulting in no
media exchange between the UAs. Older UAs may set the connection
address to 0.0.0.0 when initiating hold. However, this behavior has
been deprecated in favor of using the a=sendonly SDP attribute.
Message Details
F1 INVITE Alice -> Proxy 1
INVITE sips:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice ;tag=1234567
To: Bob
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Contact:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: ...
v=0
o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com
s=Session SDP
c=IN IP4 client.atlanta.example.com
t=3034423619 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F2 INVITE Proxy 1 -> Bob
INVITE sips:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/TLS ss1.example.com:5061;branch=z9hG4bK83749.1
Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.103
Record-Route:
Max-Forwards: 69
From: Alice ;tag=1234567
To: Bob
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Contact:
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Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: ...
v=0
o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com
s=Session SDP
c=IN IP4 client.atlanta.example.com
t=3034423619 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F3 (100 Trying) Proxy 1 -> Alice
SIP/2.0 100 Trying
Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.103
From: Alice ;tag=1234567
To: Bob
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Content-Length: 0
F4 180 Ringing Bob -> Proxy 1
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS ss1.example.com:5061;branch=z9hG4bK83749.1
;received=192.0.2.54
Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.103
Record-Route:
From: Alice ;tag=1234567
To: Bob ;tag=314159
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Contact:
Content Length:0
F5 180 Ringing Proxy 1 -> Alice
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.103
Record-Route:
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From: Alice ;tag=1234567
To: Bob ;tag=314159
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Contact:
Content Length: 0
F6 200 OK Bob -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.example.com:5061;branch=z9hG4bK83749.1
;received=192.0.2.54
Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.103
Record-Route:
From: Alice ;tag=1234567
To: Bob ;tag=314159
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Contact:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: ...
v=0
o=bob 2890844527 2890844527 IN IP4 client.biloxi.example.com
s=Session SDP
c=IN IP4 client.biloxi.example.com
t=3034423619 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F7 200 OK Proxy 1 -> Alice
SIP/2.0 200 OK
Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.103
Record-Route:
From: Alice ;tag=1234567
To: Bob ;tag=314159
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Contact:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
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Content-Type: application/sdp
Content-Length: ...
v=0
o=bob 2890844527 2890844527 IN IP4 client.biloxi.example.com
s=Session SDP
c=IN IP4 client.biloxi.example.com
t=3034423619 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F8 ACK Alice -> Proxy 1
ACK sips:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf92
Route:
Max-Forwards: 70
From: Alice ;tag=1234567
To: Bob ;tag=314159
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Length: 0
F9 ACK Proxy 1 -> Bob
ACK sips:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/TLS ss1.example.com:5061;branch=z9hG4bK837492.1
Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf92
;received=192.0.2.103
Max-Forwards: 69
From: Alice ;tag=1234567
To: Bob ;tag=314159
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Length: 0
/* Bob places Alice on hold. Note that the version is
incremented in the o= field of the SDP */
F10 INVITE Bob -> Proxy 1
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INVITE sips:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/TLS client.biloxi.example.com:5061;branch=z9hG4bKnashds7
Route:
Max-Forwards: 70
From: Bob ;tag=314159
To: Alice ;tag=1234567
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Contact:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: ...
v=0
o=bob 2890844527 2890844528 IN IP4 client.biloxi.example.com
s=Session SDP
c=IN IP4 client.biloxi.example.com
t=3034423619 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendonly
F11 INVITE Proxy 1 -> Alice
INVITE sips:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/TLS ss1.example.com:5061;branch=z9hG4bK83749.1
Via: SIP/2.0/TLS client.biloxi.example.com:5061;branch=z9hG4bKnashds7
;received=192.0.2.105
Record-Route:
Max-Forwards: 69
From: Bob ;tag=314159
To: Alice ;tag=1234567
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Contact:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: ...
v=0
o=bob 2890844527 2890844528 IN IP4 client.biloxi.example.com
s=Session SDP
c=IN IP4 client.biloxi.example.com
=== message truncated ===
---------------------------------
Jiyo cricket on Yahoo! India cricket
Yahoo! Messenger Mobile Stay in touch with your buddies all the time.
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