VoIP IP telephony cookbook available

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[An excellent resource for those interested in building VoIP networks,
particularly of you have your own wide area fiber network where you can
avoid expensive commercial VoIP services --BSA]


The IP Telephony Cookbook has been published as a TERENA Report and is
available in printed form or in pdf form from the TERENA website.

For more information:

http://www.terena.nl/news/2004/newsflash143.pdf

Carol de Groot
TERENA PR & Conference Officer
 
Table of Contents
   
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) 

 

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