Internet DRAFT - draft-donovan-sip-gw-client

draft-donovan-sip-gw-client



MMUSIC WG                                            Steve Donovan
Internet Draft                                       Matthew Cannon
                                                     MCIWorldcom
draft-donovan-sip-gw-client-00.txt
November 15, 1998
Expires: May 15, 1999


              A Functional Description of a SIP-PSTN Gateway

STATUS OF THIS MEMO

This document is an Internet-Draft. Internet-Drafts are working docu-
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1 Abstract

This document describes a functional model of a gateway between the
Public Switched Telephony Network (PSTN) and a SIP-based IP network
providing telephony service.  In this document, the SIP-based IP
network will be referred to as Telephony over IP Service (TIPS).

The primary function of the gateway is to handle connections involving 
one or more PSTN phones (i.e.; PSTN black phones) as if those 
PSTN phones are SIP clients.  The goal of this approach is to ensure 
that SIP-based TIPS service logic applies to both native Internet-
based SIP clients and PSTN phones without modification.

2 Introduction

There is currently significant effort being put into defining how the 
existing PSTN networks will interwork with the Internet.  The current 
focus of these efforts is to define how voice-based services will 
work between these networks.

There are two primary approaches emerging in this definition.  The 
first attempts to model the Internet-based telephony solutions on the 
existing PSTN architecture.  With this approach, all Internet-based 
telephone devices would be put under the control of PSTN like control 
entities.  While this approach will likely work, as it builds on what 

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has proven to be successful, albeit with limitations, from the PSTN 
networks, it unnecessarily constrains the solutions that can be 
implemented on the TIPS.

The incredible growth of the WWW argues for an approach that is not 
modeled on PSTN-like hierarchical control elements but rather on a 
distributed model similar to generic WEB services.

It is this model that was the genesis of the Session Initiation 
Protocol, which is built on the HTTP protocol.

Using this model, all users of the TIPS are modeled as SIP clients.
This includes clients that access the TIPS from PSTN phones.  This
requires the gateway function that implements the interworking 
between the PSTN and the TIPS to provide SIP client functions for all 
calls passing through the gateway.

Based on this architectural direction, this paper proposes both a 
functional decomposition and network model for the reference network 
elements defined in draft-greene-ss7-arch-frame-01.txt.

3 Gateway Functional Elements

The following is a description of the functional elements of the PSTN 
to SIP TIPS gateway.

Signaling Gateway - This function terminates the signaling associated 
with a given PSTN channel/circuit.  There are multiple types of SGs in 
this model.  The following is a partial list of such Signaling 
Gateways:

SS7 Signaling Gateway - This SG type terminates call associated SS7 
signaling.  One example of the SS7 signaling is ISUP.  Note that in 
this model the SG does not handle TCAP messaging, although a TCAP SG 
could easily be added to the model.

ISDN Signaling Gateway - This SG type terminates ISDN call associated 
signaling.

CAS Signaling Gateway - This SG type terminates signaling associated 
with CAS circuits.  Examples include MF and R2-based circuits.

Media Gateway - This function is very similar to that proposed in the
SS7 architecture framework.  This function will implement the media 
level interworking between the PSTN circuit and an RTP flow.

SIP Client - This function will handle the session initiation 
signaling on the IP side of the gateway.

RTP Client - This function will handle the RTP flow.



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Media Control - In this model the function of Media Control is split 
between the gateway and a SIP server.  However, the SIP server 
function is not required in this model, in which case the full MC 
function would be imbedded in the gateway.

4 Gateway Types

Although the functions of the gateway do not depend on the type of PSTN
signaling used, the following functional models of the gateways are 
provided for illustrative purposes.

Note that this description of gateway types does not assume that they 
would be deployed separately.  It is expected that a single gateway 
instance would be able to handle multiple PSTN signaling types.

4.1 ISDN Gateway

An ISDN Gateway will handle the interface to ISDN-based PSTN trunks and
lines.
 
           +-----------------------+
           |                       |
           | +------+   +--------+ |          +--------+ 
 D-Channel | | ISDN |   |  SIP   | |  SIP     |  SIP   |
<----------->|  SG  |<->| Client |<---------->| Server |
           | +------+   +--------+ |          |   or   |
           |    ^                  |          | Client |
           |    |                  |          +--------+
           |    v                  |
           | +------+   +--------+ |          +--------+
 B-Channel | | ISDN |   |  RTP   | |  RTP     |  SIP   |
<--------->| |  MG  |<->| Client |<---------->| Client |
           | +------+   +--------+ |          |        |
           |                       |          +--------+
           +-----------------------+


















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4.2 CAS Gateway

A Channel Assocated Signaling Gateway will handle the interface to CAS-
based PSTN trunks and lines.

           +-----------------------+
           |                       |
           | +------+   +--------+ |          +--------+
           | |  CAS |   |  SIP   | |  SIP     |  SIP   |
           | |  SG  |<->| Client |<---------->| Server |
           | +------+   +--------+ |          |   or   |
           |    ^                  |          | Client |
           |    |                  |          +--------+
           |    v                  |
           | +------+   +--------+ |          +--------+
 Circuit   | |  CAS |   |  RTP   | |  RTP     |  SIP   |
<--------->| |  MG  |<->| Client |<---------->| Client |
           | +------+   +--------+ |          |        |
           |                       |          +--------+
           +-----------------------+

4.3 SS7 Gateway

A SS7 Gateway will handle the interface to PSTN trunks that are 
controlled by ISUP, TUP or other SS7-based call signaling methods.  

This type of gateway is differentiated from the ISDN and CAS gateways
by the fact that it must be implemented in a fashion that allows the 
SG function can be deployed separately from the MG function.  This is
due to the relative expense of the SS7 link and the scarcity of SS7 
point codes.

           +-----------------------+
           |                       |
           | +------+   +--------+ |          +--------+
 SS7 Link  | |  SS7 |   |  SIP   | |  SIP     |  SIP   |
<----------->|  SG  |<->| Client |<---------->| Server |
           | +------+   +--------+ |          |   or   |
           |    ^                  |          | Client |
           |    |                  |          +--------+
           +-----------------------+
                |                  
                |  
           +-----------------------+
           |    |                  |
           |    v                  |
           | +------+   +--------+ |          +--------+
 Circuit   | |  SS7 |   |  RTP   | |  RTP     |  SIP   |
<--------->| |  MG  |<->| Client |<---------->| Client |
           | +------+   +--------+ |          |        |
           |                       |          +--------+
           +-----------------------+

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5 Interfaces

The definition of the internal interfaces is outside of the scope of 
this document and is assumed to be an implementation issue.

It is important that all other interfaces be standardized.  The SIP and
RTP standards are already in various stages of standardization.

The interface between the SS7 SG and the SS7 MG will require 
standardization.  The current work on the MGCP protocol is one 
candidate for this interface.  Another option is extensions to the SIP
protocol.

6 DTMF Handling

An important function of the gateway is to convey all signaling 
information received from the PSTN to the TIPS.  In all three gateway 
types defined, signaling information can occur as DTMF digits encoded 
into the voice stream.  

This "inband" call control information can be used for communications 
between the PSTN user and another end device, for example a service 
node or the user's at-home answering machine.  In this case, it is 
envisioned that the DTMF information will be carried in the TIPS as 
part of the RTP stream.

This call control information can also be used for interactions with 
service control logic generally deployed in the PSTN in Service Control
Points.  It is envisioned that the same method of communication between
PSTN-based clients of the TIPS and the TIPS-based service control will 
be required.  As such, it is important that the gateway be able to 
detect and report on the presence of DTMF digits in the voice stream.  
Note that it is expected that a similar method of communicating between 
native SIP clients and TIPS service logic will be required for SIP 
clients with nothing more than a telephone keypad available to control 
service logic.

There is work currently underway to define extensions to the SIP 
protocol that will allow for an interested IP host, for instance a SIP 
server, to request notification of call events.  One such use of this 
extension will be to request notification of the detection of DTMF 
digits.  This would allow the SIP server, or other interested host, 
to use the DTMF digits for service control logic.

7 Conclusion

It is important that the definition of the interface between the PSTN 
and the Internet for telephony-based services take advantage of the 
richness of services made possible by the Internet and the World Wide 
Web.  In order to ensure that this is possible, it is necessary to 
model the telephony services on the Internet first and then determine
how to interwork with the PSTN.  

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This paper proposes doing this by treating all PSTN devices as SIP 
clients.  In doing so, services developed for native Internet SIP 
clients can be offered to PSTN clients without modification.  In 
addition, new services can be added to the TIPS with the relative ease 
of adding a WEB page.

8 Bibliography

[1] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, 
    "SIP: Session Initiation Protocol", Internet Draft, Internet 
    Engineering Task Force, October 14, 1998.  Work in progress.

[2] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
    Transport Protocol for Real-Time Applications", RFC 1889, January
    1996.

[3] F. Cuervo, N. Greene, M. Holdrege, L. Ong, C. Huitema, "SS7-
    Internet Interworking - Architectural Framework", Internet
    Draft, Internet Engineering Task Force, July, 1998.

9 Author's Address

   Steve Donovan
   MCI Worldcom
   1493/678
   901 International Parkway
   Richardson, Texas 75081
   Email: steven.r.donovan@mci.com

   Matthew Cannon
   MCI Worldcom
   9514/107
   2400 North Glenville Drive
   Richardson, Texas 75082
   Email: matt.cannon@mci.com 


















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