IP Telephony Cookbook Engineering Telecommunication Engineering

INTRODUCTION:Goal, Reasons for writing this document, How to read this document >>

IP Telephony Cookbook

> TERENA REPORT //

> / MARCH, 2004

The IP Telephony Cookbook was created through the IP Telephony project as a reference

document for setting up IP Telephony solutions at university campuses and NRENs.

The project started in Apr il 2003 and ran until February 2004. The Cookbook provides an

overview of available and future IP Telephony technologies, scenar ios for IP Telephony

deployment and infrastructures, guidelines on protocols, service set-ups and connection to a global

`dialling plan'. Further more, the Cookbook reports on the interoperability of

equipment, existing IP Telephony projects and regulatory aspects.

The project was car r ied out by the University of Pisa, Italy, TZI-University of Bremen

and FhG FOKUS, Ger many, with contr ibutions from CESNET, GRNET, SURFnet and

the University of Graz.

Funding to the project was provided by TERENA, ARNES, CARNet, CYNET, SUNET and

UKERNA. Representatives of the funding organisations were members of the Project Review

Committee.

ISBN 90-7759-08-6

Parts of this report may be freely copied, unaltered, provided that the original source is

acknowledged and the copyright preserved.

Production: TERENA Secretariat

Design: Eva de Lange

Printing: GraphicResult

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[IP Telephony Cookbook] / Contents

AUTHORS :

Margit Brandl - Karl Franzens - UNI Graz, Dimitris Daskopoulos - GRNET,

Erik Dobbelsteijn - SURFnet, Rosario Giuseppe Garroppo - University of Pisa, Jan Janak - FhG Fokus,

Jiri Kuthan - FhG Fokus, Saverio Niccolini - University of Pisa, Jörg Ott - Universität Bremen TZI,

Stefan Prelle - Universität Bremen TZI, Sven Ubik - CESNET Uni Graz, Egon Verharen - SURFnet

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[IP Telephony Cookbook] / Contents

CONTENTS //

1. INTRODUCTION

1.1

Goal

1.2

Reasons for writing this document

1.3

Contents

1.4

How to read this document

1.5

Techno-economic aspects of moving from classic telephony to VoIP

2. TECHNOLOGICAL BACKGROUND

2.1 Components

2.1.1 Terminal

2.1.2 Server

2.1.3 Gateway

2.1.4 Conference bridge

2.1.5 Addressing

2.2 Protocols

2.2.1 H.323

2.2.2 SIP

2.2.3 Media gateway control protocols

2.2.4 Proprietary signalling protocols

2.2.5 Real Time Protocol (RTP) and Real Time Control Protocol (RTCP)

3. IP TELEPHONY SCENARIOS

3.1 Introduction

3.2 Scenario 1: Long-distance least cost routing

3.2.1 Least cost routing - an example of an implementation

3.3 Scenario 2: Alternatives to legacy PBX systems

3.3.1 Scenario 2a: IP Phones without a PBX system

3.3.2 Scenario 2b: Integration of VoIP with legacy PBX systems

3.3.3 Scenario 2c: Full replacement of legacy PBX systems

3.4 Scenario 3: Integration of VoIP and videoconferencing

3.4.1 Integrating voice and videoconferencing over IP - an example

4. SETTING UP BASIC SERVICES

4.1 General concepts

4.1.1 Architecture

4.1.2 Robustness

4.1.3 Management issues

4.2. Dial plans

4.3 Authentication

4.3.1 Authentication in H.323

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[IP Telephony Cookbook] / Contents

4.3.2 Authentication in SIP

4.4

Examples

4.4.1 Example 1: simple, use IP Telephony like legacy telephony

4.4.2 Example 2: complex, full-featured

4.5

Setting up H.323 services

4.5.1 Using a Cisco Multimedia Conference Manager (MCM Gatekeeper)

4.5.2 Using a RADVI SION-Enhanced Communication Server (ECS Gatekeeper)

4.5.3 Using an OpenH.323 Gatekeeper - GNU Gatekeeper

4.6

Setting up SIP services

4.6.1 Operation of SIP servers

4.6.2 SIP express router

4.6.3 Asterisk

4.6.4 VOCAL

4.7

Firewalls and NAT

4.7.1 Firewalls and IP Telephony

4.7.2 NAT and IP Telephony

4.7.3 SIP and NAT

5. SETTING UP ADVANCED SERVICES

135

5.1 Gatewaying

5.1.1 Gateway interfaces

5.1.2 Gatewaying from H.323 to PSTN/ISDN

5.1.3 Gatewaying from SIP to PSTN/ISDN

5.1.4 Gatewaying from SIP to H.323 and vice versa

5.1.5 Accounting gateways

5.2 Supplementary services

5.2.1 Supplementary services using H.323

5.2.2 Supplementary services using SIP

5.3 Multipoint conferencing

6. SETTING UP VALUE-ADDED SERVICE

173

6.1 Web integration of H.323 services

6.1.1 RADIUS-based methods

6.1.2 SNMP-based methods

6.1.3 Cisco MCM GK API

6.1.4 GNU GK status interface

6.2 Web integration of SIP services

6.2.1 Click-to-dial

6.2.2 Presence

6.2.3 Missed calls

6.2.4 Serweb

6.2.5 SIP express router message store

6.3 Voicemail

7. INTEGRATION OF GLOBAL TELEPHONY

184

7.1 Technology

7.1.1 H.323 LRQ

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7.1.2 H.225.0 Annex G

7.1.3 Telephony routing over IP (TRIP)

7.1.4 SRV-records

7.1.5 ENUM

7.2 Call routing today

7.2.1 SIP

7.2.2 Using H.323

7.3 Utopia: setting up global IP Telephony

7.4 Towards Utopia

7.4.1 Call routing assistant

8. REGULATORY / LEGAL CONSIDERATIONS

200

8.1 Overall

8.2What does regulation mean for Voice over IP?

8.3 Regulation of Voice over IP in the European Union

8.3.1 Looking back into Europe's recent history in regulation

8.3.2 The new regulatory framework - technological neutrality

8.3.3 The new regulatory framework - an overview

8.3.4 Authorisation system instead of licensing system

8.3.5 Numbering

8.3.6 Access

8.3.7 Interconnection

8.3.8 Quality of Service

8.4Voice over IP in the United States

8.5Conclusion and summary

// ANNEX A. European IP Telephony Projects

210

A.1Evolute

A.26Net

A.3Eurescom P1111 (Next-Gen open Service Solutions over IP (N-GOSSIP)

A.4HITEC

A.5The GRNET/RTS project

A.6SURFWorks

A.7VC Stroom

A.8Voice services in the CESNET2 network

// ANNEX B. IP Telephony Hardware/Software

214

B.1Softphones

B.2Hardphones

B.3Servers

B.4Gateways

B.5Testing

B.6Miscellaneous

// GLOSSARY

226

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