One document matched: draft-rosenberg-sip-3pcc-00.txt
Internet Engineering Task Force SIP WG
Internet Draft J.Rosenberg,J.Peterson,H.Schulzrinne
draft-rosenberg-sip-3pcc-00.txt dynamicsoft,Level3,Columbia U.
March 10, 2000
Expires: September, 2000
Third Party Call Control in SIP
STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
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Abstract
This document discusses the usage of SIP for third party call
control. Third party call control refers to the ability of one entity
to create a call in which communications is actually between other
parties. We present a SIP mechanism for accomplishing third party
call control that does not require any extensions or changes to SIP.
1 Introduction
In the traditional telephony context, third party call control refers
to the ability of one entity (which we call the controller), to
create, modify, or terminate calls between other participants. Third
party call control is often used for operator services (where an
operator creates a call that connects two participants together), and
conferencing.
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On the Internet, a wider range of services are enabled through a
third party session control mechanism. This is because other IP
applications, such as web, email, presence, instant messaging, and
chat can now be brought into the picture. An excellent example is
click-to-dial. This service allows a user to click on a web page when
they wish to speak to a customer service representative. The web
server then creates a call between the user and a customer service
representative. The call can be between two phones, a phone and an IP
host, or two IP hosts.
In order to support third party call control applications, a
mechanism is needed that allows a controller to create, modify, and
terminate calls with other entities. In this document, we present a
mechanism using SIP which allows a controller to execute third party
services. The mechanism is not an extension to SIP. It is merely an
application of the tools enabled through RFC 2543. A controller can
create calls between any entity that contains a normal SIP user
agent. After desribing the mechanism, we present three third party
services which take advtantage of this mechanism. One is click-to-
dial, the second is a feature that enables a mid-call announcement,
and the third is a timed conference bridge initiation.
2 Third Party Control
The basic idea behind the third party mechanism is simple. Consider
first the case of just connecting two users in a call. The controller
first sends an INVITE to the first user whose phone is to ring. This
is a standard INVITE, but it contains no SDP.[1] send an ACK. It
generates a second INVITE. This INVITE is addressed to the second
user to be connected in the call. This INVITE contains the SDP as
received from the 200 OK of the first user. When the 200 OK to this
second INVITE arrives, the controller ACK s it, takes the SDP, and
includes that in the ACK for the first call. A flow diagram for this
mechanism is given in Figure 1.
| INV no SDP | |
|<------------------| |
| | |
| 200 SDP A | |
|-----------------> | INV SDP A |
_________________________
[1] RFC 2543 does allow for the initial INVITE to not
contain a session description
J.Rosenberg,J.Peterson,H.Schulzrinne [Page 2]
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| |----------------->|
| | |
| | 200 SDP B |
| |<-----------------|
| | |
| | ACK |
| ACK SDP B |----------------->|
|<------------------| |
| | |
| | RTP |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
A Controller B
Figure 1
At this point, both participants believe they are in a single point
to point call with some control system (assuming the controller
identified itself as such in the From field of the INVITE). However,
they are exchaning media directly with each other, rather than with
the controller. The result is that the controller has set up a call
between both participants.
Since the controller is still a central point for signaling, it now
has complete control over the call. If it receives a BYE from one of
the participants, it can create a new BYE and hang up with the other
participant. This is shown in Figure 2.
| | |
| | |
| BYE From A | |
|-----------------> | BYE From Cont. |
| 200 OK |----------------> |
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|<----------------- | 200 OK |
| |<---------------- |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
A Controller B
Figure 2
As an alternative, when the controller receives a BYE from A, it can
generate a new INVITE to a third party, C, using the SDP from B. When
the 200 OK arrives from C, the controller sends a re-INVITE to B,
using the SDP from C. If the 200 OK to the re-INVITE contains the
same SDP as it used in the INVITE to C, the controller has
sucessfully connected B to C, transparently to B. A call flow for
this is shown in Figure 3.
| | | |
| | | |
| BYE From A | | |
|-----------------> | INV SDP B | |
| 200 OK |------------------------------------>|
|<----------------- | | 200 SDP C |
| |<------------------------------------|
| | ACK | |
| |------------------------------------>|
| | INV SDP C | |
| |----------------->| |
| | 200 SDP B | |
| |<-----------------| |
| | ACK | |
| |----------------->| |
| | | |
| | | RTP |
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| | xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx |
| | xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx |
| | | |
| | | |
| | | |
| | | |
A Controller B C
From here, new parties can be added, removed, transferred, and so on,
as the controller sees fit.
The general idea behind the mechanism is that there is a point to
point SIP relationship between each participant and the controller.
However, by passing the SDP it receives from one participant to
another, it can causes users to actually communicate with each other
rather than the controller.
3 Click to Dial
The first application of this capability we discuss is click to dial.
In this service, a user is browsing the web page of an e-commerce
site, and would like to speak to a customer service representative.
They click on a link, and the phone on the desk (a normal telephone)
rings. When the user picks up, the phone of the customer service
representative (an IP phone) rings. When they pick up, the service
representative is talking to the user.
We assume for purposes of this discussion that the web server is
actually an applications server that contains an http interface. In
this case, when the user clicks on the URL, the application server
knows, through cookies or some other state mechanism, the addresses
of the participants to be connected.
The call flow for this service is given in Figure 4. Note that it is
identical to that of Figure 1, with the exception that the service is
triggered through an http GET request when the user clicks on the
link.
| | HTTP GET | |
| |<-----------------+-------------------|
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| | 200 OK | |
| |------------------+------------------>|
| | | |
| | | |
| | | |
| | | |
| INV no SDP | | |
|<------------------| | |
| | | |
| 200 SDP A | | |
|-----------------> | INV SDP A | |
| |----------------->| |
| | | |
| | 200 SDP B | |
| |<-----------------| |
| | | |
| | ACK | |
| ACK SDP B |----------------->| |
|<------------------| | |
| | | |
| | RTP | |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx | |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
PSTN Controller Customer Users PC
GW Service
Representative
Figure 4
We note that this service can be provided through other mechanisms,
namely PINT [1]. However, there are numerous differences between the
way in which the service is provided by pint, and the way in which it
is provided here:
o The pint solution enables calls only between two PSTN
endpoints. The solution described here allows calls between
PSTN phones (through SIP enabled gateways) and native IP
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phones.
o When used for calls between two PSTN phones, the solution here
may result in a portion of the call being routed over the
Internet. In pint, the call is always routed only over the
PSTN. This will probably result in better quality calls with
the pint solution.
o The PINT solution requires extensions to SIP (PINT is an
extension to SIP), whereas the solution described here is done
with baseline SIP.
o The PINT solution allows the controller (acting as a PINT
client) to "step out" once the call is established. The
solution described here requires the controller to remain in
the picture for the entire duration of the call.
4 Mid-Call Announcement Capability
The third party call control mechanism described here can also be
used to enable mid-call announcements. The call is set up by the
controller as desribed above.[2] payphone, in which case the
controller determines that the call is to be terminated after some
amount of time if the user doesn't add more money to the phone. When
this timer expires, the controller initiates places the called party
on hold. It then sends an INVITE to the media server which will be
collecting digits. It then sends a re-INVITE to the user on the
payphone, connecting its media streams with the media server. The
media server plays an announcement, and prompts the user to enter a
credit card number, for example. After collecting the number and
validating the card, if the call can continue, the media server hangs
up. The controller takes this as a cue and reconnects the user to the
original called party, and takes the original called party off hold.
A call flow for this service is shown in Figure 5
| RTP | | |
_________________________
[2] It is actually not necessary for the controller
to set up the call. However, if a participant initiates
the call, the controller must step in as a virtual
UAC/UAS, and act as a termination and re-initiation
point
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|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx| |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx| |
| | INV SDP 0 (hold) | |
| |----------------->| |
| | 200 OK | |
| |<-----------------| |
| | ACK | |
| |----------------->| |
| | INV SDP A | |
| |------------------------------------->|
| | | 200 SDP C |
| INV SDP C |<-------------------------------------|
|<------------------| ACK | |
| 200 SDP A |------------------------------------->|
|------------------>| | |
| ACK | | |
|<------------------| | |
| | | |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx|
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx|
| | | BYE |
| |<-------------------------------------|
| | 200 OK | |
| |------------------------------------->|
| | INV SDP A | |
| |----------------->| |
| | 200 SDP B | |
| INV SDP B |<-----------------| |
|<------------------| ACK | |
| 200 SDP A |----------------->| |
|------------------>| | |
| ACK | | |
|<------------------| | |
| | | |
Payphone Controller Called Media
"A" Party Server
"B" "C"
Figure 5
We have assumed that the media server and the controller have agreed,
ahead of time, that a hangup implies that the desired service
(extending the lifetime of the call) has succeeded. This is
effectively allowing a call control interface between the controller
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and the media server. Parameters needed between the elements, such as
the new expiration of the call, can be passed in the BYE. A separate
draft, forthcoming, will discuss call control interfaces to media
services in more detail.
5 Timed Conference Intitation
In this service, a conference bridge is booked for some number of
participants. In order to make sure the conference begins on time,
the conference bridge will call each participant at the time of the
call. If a participant doesn't answer, the bridge tries to contact
them again (unless they call in) five minutes later.
In the call flow described here, we assume that the controller acts
as the media bridge. This is not strictly necessary; some kind of
control interface could be used to separate the media function from
the controller.
The call flow, shown in Figure 6, is, not surprisingly, remarkably
like that of Figure 1. The only difference is that the SDP listed in
the INVITE s generated by the controller always contain SDP that
points to the conference bridge, rather than one of the other
participants. In the call flow diagram, user 1 is invited first, then
user 2, and then user 3. User 3 is not available, but is called again
five minutes later.
| INV SDP X | | |
|<------------------| | |
| | INV SDP X | |
| 200 SDP A |----------------->| |
|-----------------> | | |
| | 200 SDP B | |
| ACK |<-----------------| |
|<------------------| | |
| | ACK | |
| |----------------->| |
| | | |
| | INV SDP X | |
| |--------------------------------------->|
| | | 408 Timeout |
| |<---------------------------------------|
| | ACK | |
| |--------------------------------------->|
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| | | |
| | | |
| | INV SDP X | |
| |--------------------------------------->|
| | | 408 Timeout |
| |<---------------------------------------|
| | ACK | |
| |--------------------------------------->|
| | | |
| | | |
| | | |
| | | |
| | | |
User 1 Controller User 2 User 3
"A" "X" "B" "C"
Figure 6
6 Future Work
We plan on considering other mechanisms which might be used for third
party call control, and discuss the pros and cons for each for
providing numerous services.
7 Conclusions
We have presented a basic third party call control mechanism that
uses SIP. This mechanism does not require any extensions to SIP and
is completely backwards compatible.
8 Authors Addresses
Jonathan Rosenberg
dynamicsoft
200 Executive Drive
Suite 120
West Orange, NJ 07052
email: jdrosen@dynamicsoft.com
Jon Peterson
Level 3 Communications
Henning Schulzrinne
Columbia University
J.Rosenberg,J.Peterson,H.Schulzrinne [Page 10]
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M/S 0401
1214 Amsterdam Ave.
New York, NY 10027-7003
email: schulzrinne@cs.columbia.edu
9 Bibliography
[1] S. Petrack and L. Conroy, "The PINT service protocol: Extensions
to SIP and SDP for IP access to telephone call services," Internet
Draft, Internet Engineering Task Force, June 1999. Work in progress.
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