IP Telephony Cookbook Dr. Margit Brandl, Dimitris Daskopoulos, Erik Dobbelsteijn, Dr. Rosario Giuseppe Garroppo, Jan Janak, Jiri Kuthan, Saverio Niccolini, Dr. Jorg¨ Ott, Stefan Prelle, Dr. Sven Ubik, Egon Verharen March 9, 2004 Contents 1 Introduction 11 1.1 Goal . 11 1.2 Reasons for writing this document . 11 1.3 Contents . 11 1.4 How to read this document . 12 1.5 Techno-economic aspect of moving from classic telephony to VoIP . 12 2 Technology Background 15 2.1 Components . 15 2.1.1 Terminal . 15 2.1.2 Server . 15 2.1.3 Gateway . 15 2.1.4 Conference Bridge . 16 2.1.5 Addressing . 16 2.2 Protocols . 16 2.2.1 H.323 . 16 2.2.1.1 Scope . 17 2.2.1.2 Signaling protocols . 18 2.2.1.3 Gatekeeper Discovery and Registration . 20 2.2.1.4 Signaling models . 21 2.2.1.5 Communication Phases . 24 2.2.1.6 Locating zone external targets . 28 2.2.1.7 Sample Call Scenario . 29 2.2.1.8 Additional (Call) Services . 29 2.2.1.9 H.235 Security . 30 2.2.1.10 Protocol Profiles . 31 2.2.2 SIP . 31 2.2.2.1 Purpose of SIP . 31 2.2.2.2 SIP Network Elements . 32 2.2.2.3 SIP Messages . 34 2.2.2.4 SIP Transactions . 38 2.2.2.5 SIP Dialogs . 38 2.2.2.6 Typical SIP Scenarios . 40 2.2.3 Media Gateway Control Protocols . 43 2.2.4 Proprietary Signaling Protocols . 45 2.2.5 Real Time Protocol (RTP) and Real Time Control Protocol (RTCP) . 45 2.2.5.1 RTP Header . 46 2.2.5.2 RTCP packet types and format . 47 3 IP Telephony Scenarios 49 3.1 Introduction . 49 3.2 Scenario 1: Long-distance least cost routing . 49 3.2.1 Least Cost Routing - An implementation example . 50 3.3 Scenario 2: Alternatives to legacy PBX systems . 50 3.3.1 Scenario 2a: IP-Phones without a PBX system . 51 3.3.2 Scenario 2b: Integration of VoIP with legacy PBX systems . 52 3.3.3 Scenario 2c: Full replacement of legacy PBX systems . 53 3.3.3.1 Intelligent vs. simple terminals . 53 3.3.3.2 Signalling . 54 3.3.3.3 Inter-department trunking . 54 3.3.3.4 Legacy functionality . 54 3.3.3.5 Wireless VoIP . 55 3.3.3.6 Issues . 55 3.4 Scenario 3: Integration of VoIP and Videoconferencing . 55 3 CONTENTS 3.4.1 Integrating Voice and Videoconferencing over IP - an example . 56 4 Setting up basic services 59 4.1 General concepts . 59 4.1.1 Architecture . 59 4.1.1.1 PSTN gateways / PBX migration . 60 4.1.1.2 Trunking . 62 4.1.2 Robustness . 63 4.1.3 Management issues . 65 4.1.3.1 Multiple account databases . 65 4.1.3.2 Decentralization . 66 4.2 Dialplans . 66 4.3 Authentication . 67 4.3.1 Authentication in H.323 . 67 4.3.1.1 Areas of application . 67 4.3.1.2 User Authentication . 67 4.3.1.3 Integrity . 68 4.3.1.4 Confidentiality . 68 4.3.1.5 Security profiles . 68 4.3.1.6 H.235 and the real world . 69 4.3.2 Authentication in SIP . 69 4.3.2.1 Overview of Digest Authentication . 69 4.3.2.2 Digest Authentication and SIP . 70 4.3.2.3 Basic Scenarios . 71 4.4 Examples . 73 4.4.1 Example 1: Simple, use IP telephony like legacy telephony . 73 4.4.2 Example 2: Complex, full featured . 73 4.5 Setting up H.323 services . 75 4.5.1 Using a Cisco Multimedia Conference Manager (MCM gatekeeper) . 75 4.5.1.1 Installation . 75 4.5.1.2 Configuration . 76 4.5.1.3 Operation . 77 4.5.1.4 Endpoint authentication . 78 4.5.1.5 Advanced features . 78 4.5.2 Using a Radvision Enhanced Communication Server (ECS gatekeeper) . 78 4.5.2.1 Installation . 79 4.5.2.2 Configuration . 79 4.5.2.3 Operation . 81 4.5.2.4 Endpoint authentication . 81 4.5.2.5 Advanced features . 81 4.5.3 Using an OpenH323 Gatekeeper - GNU Gatekeeper . 82 4.5.3.1 Installation . 82 4.5.3.2 Configuration . 84 4.5.3.3 Operation . 85 4.5.3.4 Endpoint authentication . 85 4.5.3.5 Advanced features . 86 4.6 Setting up SIP services . 86 4.6.1 Operation of SIP Servers . 86 4.6.1.1 Recommended Operational Practices . 86.