IP Telephony Cookbook

Download entire publication (pdf - 7,761 kb)

TABLE OF CONTENTS

1. INTRODUCTION (pdf - 42 kb)
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 (pdf - 533 kb)
2.1 Components
- 2.1.2 Terminal
- 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 (pdf - 1072 kb)
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 (pdf - 1278 kb)
4.1 General concepts
- 4.1.1 Architecture
- 4.1.2 Robustness
- 4.1.3 Management issuese
4.2 Dial plans
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 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 (pdf - 2475 kb)
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 (pdf - 1587 kb)
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 (pdf - 790 kb)
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 (pdf - 72 kb)
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 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
8.4 Voice over IP in the United States
8.5 Conclusion and summary
ANNEX A - European IP Telephony Projects (pdf - 262 kb)
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
ANNEX B - IP Telephony Hardware/Software (pdf - 62 kb)
B.1 Softphones
B.2 Hardphones
B.3 Servers
B.4 Gateways
B.5 Testing
B.6 Miscellaneous
GLOSSARY (pdf - 36 kb)