IP Telephony Cookbook

Table of 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 aspect of moving from classic telephony to VoIP

2. Technology 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 Signaling 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 implementation example

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. Dialplans

4.3. Authentication

4.3.1. Authentication in H.323

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 Radvision Enhanced Communication Server (ECS gatekeeper)

4.5.3. Using an OpenH323 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

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 Services

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. Global telephony integration

7.1. Technology

7.1.1. H.323 LRQ

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

8.1. Overall

8.2. What 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. New Regulatory Framework - an overview

8.3.4. Authorization System instead of Licensing System

8.3.5. Numbering

8.3.6. Access

8.3.7. Interconnection

8.3.8. Quality of Service

8.4. Voice over IP in the United States

8.5. Conclusion and Summary

A. European IP Telephony Projects

A.1. Evolute

A.2. 6Net

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

A.4. HITEC

A.5. The GRNET/RTS project

A.6. SURFWorks

A.7. VC Stroom

A.8. Voice services in the CESNET2 network

B. IP Telephony Hardware/Software

B.1. Softphones

B.2. Hardphones

B.3. Servers

B.4. Gateways

B.5. Testing

B.6. Miscellaneous

Glossary