| 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) |
