IP Telephony Cookbook Deliverable 2 Saverio Niccolini, University of Pisa Dr. Rosario Giuseppe Garroppo, University of Pisa Dr. Jörg Ott, Universität Bremen TZI Stefan Prelle, Universität Bremen TZI Jiri Kuthan, FhG Fokus Jan Janak, FhG Fokus Dr. Sven Ubik, CESNET Dr. Margit Brandl, Karl-Franzens-Uni Graz Dimitris Daskopoulos, GRNET Egon Verharen, SURFnet Erik Dobbelsteijn, SURFnet IP Telephony Cookbook: Deliverable 2 by Saverio Niccolini, Dr. Rosario Giuseppe Garroppo, Dr. Jörg Ott, Stefan Prelle, Jiri Kuthan, Jan Janak, Dr. Sven Ubik, Dr. Margit Brandl, Dimitris Daskopoulos, Egon Verharen, and Erik Dobbelsteijn Table of Contents 1. Introduction ....................................................................................................................... 1 1.1. Goal ...................................................................................................................... 1 1.2. Reasons for writing this document ............................................................................... 1 1.3. Contents ................................................................................................................. 1 1.4. How to read this document ......................................................................................... 1 1.5. Techno-economic aspect of moving from classic telephony to VoIP ................................... 2 2. Technology Background ...................................................................................................... 5 2.1. Components ............................................................................................................ 5 2.1.1. Terminal ...................................................................................................... 5 2.1.2. Server .......................................................................................................... 5 2.1.3. Gateway ...................................................................................................... 5 2.1.4. Conference Bridge ......................................................................................... 5 2.1.5. Addressing ................................................................................................... 6 2.2. Protocols ................................................................................................................ 6 2.2.1. H.323 .......................................................................................................... 6 2.2.2. SIP ............................................................................................................ 21 2.2.3. Media Gateway Control Protocols ................................................................... 37 2.2.4. Proprietary Signaling Protocols ...................................................................... 39 2.2.5. Real Time Protocol (RTP) and Real Time Control Protocol (RTCP) ...................... 39 3. IP Telephony Scenarios ..................................................................................................... 43 3.1. Introduction .......................................................................................................... 43 3.2. Scenario 1: Long-distance least cost routing ................................................................ 43 3.2.1. Least Cost Routing - An example of integration ................................................. 45 3.3. Scenario 2: Legacy PBX replacement ......................................................................... 45 3.3.1. Scenario 2a: IP-Phone without PBX ................................................................ 45 3.3.2. Scenario 2b: Integration with legacy PBX systems ............................................. 46 3.3.3. Scenario 2c: Full replacement ........................................................................ 47 3.4. Scenario 3: Integration of VoIP and Videoconferencing ................................................. 49 3.4.1. Integrating Voice and Videoconferencing over IP - an example ............................ 50 4. Setting up basic services .................................................................................................... 55 4.1. General concepts .................................................................................................... 55 4.1.1. Architecture ................................................................................................ 55 4.1.2. Robustness ................................................................................................. 60 4.1.3. Management issues ...................................................................................... 61 4.1.4. Operation of SIP Servers ............................................................................... 62 4.2. Dialplans .............................................................................................................. 64 4.3. Authentication and Billing ....................................................................................... 65 4.3.1. Authentication in H.323 ................................................................................ 65 4.3.2. Authentication in SIP ................................................................................... 67 4.4. Examples .............................................................................................................. 71 4.4.1. Example 1: Simple, use IP telephony like legacy telephony .................................. 71 4.4.2. Example 2: Complex, full featured .................................................................. 71 4.5. Setting up H.323 services ......................................................................................... 72 4.5.1. Using a Cisco Multimedia Conference Manager (MCM gatekeeper) ...................... 73 4.5.2. Using a Radvision Enhanced Communication Server (ECS gatekeeper) ................. 76 4.5.3. Using an OpenH323 Gatekeeper - GNU Gatekeeper ........................................... 78 4.6. Setting up SIP services ............................................................................................ 82 4.6.1. SIP Express Router ...................................................................................... 83 4.6.2. Asterisk ..................................................................................................... 97 4.6.3. VOCAL ..................................................................................................... 99 4.7. Firewalls and NAT ................................................................................................102 4.7.1. Firewalls and IP telephony ............................................................................102 4.7.2. NAT and IP telephony .................................................................................102 4.7.3. SIP and NAT .............................................................................................103 5. Setting up Advanced Services ............................................................................................109 5.1. Gatewaying ..........................................................................................................109 5.1.1. Gateway interfaces ......................................................................................109 v IP Telephony Cookbook 5.1.2. Gatewaying from H.323 to PSTN/ISDN ..........................................................114 5.1.3. Gatewaying from SIP to PSTN/ISDN .............................................................120 5.1.4. Gatewaying from SIP to H.323 and vice versa ..................................................121 5.1.5. Accounting Gateways ..................................................................................124 5.2. Supplementary services ..........................................................................................124 5.2.1. Supplementary Services using H.323 ..............................................................124 5.2.2. Supplementary Services using SIP .................................................................131 5.3. Multipoint Conferencing .........................................................................................137 6. Setting up Value-Added Services ........................................................................................139 6.1. Web Integration of H.323 services ............................................................................139 6.1.1. RADIUS-based methods ..............................................................................139 6.1.2. SNMP-based methods .................................................................................139 6.1.3. Cisco MCM GK API ...................................................................................139 6.1.4. GNU GK Status Interface .............................................................................140 6.2. Web Integration of SIP Services ...............................................................................140 6.2.1. Click-to-Dial ..............................................................................................140 6.2.2. Presence ...................................................................................................142 6.2.3. Missed Calls ..............................................................................................142 6.2.4. Serweb .....................................................................................................142 6.2.5. SIP Express Router Message Store .................................................................146