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Table Of Contents

1.1 Goal
1.2 Reasons for writing this document
1.3 Contents
1.4 How to read this document
Chapter 2
Technology Background
2.1 Components
2.1.1 Terminal
2.1.2 Server
2.1.3 Gateway
2.1.4 Conference Bridge
2.1.5 Addressing
2.2 Protocols
2.2.1 H.323 Scope Signaling protocols Gatekeeper Discovery and Registration Signaling models Communication Phases Locating zone external targets Sample Call Scenario Additional (Call) Services H.235 Security Protocol Profiles
2.2.2 SIP Purpose of SIP SIP Network Elements SIP Messages SIP Transactions SIP Dialogs Typical SIP Scenarios
2.2.3 Media Gateway Control Protocols
2.2.4 Proprietary Signaling Protocols
2.2.5 Real Time Protocol (RTP) and Real Time Control Protocol (RTCP) RTP Header RTCP packet types and format
Chapter 3
IP Telephony Scenarios
3.1 Introduction
3.2 Scenario 1: Long-distance least cost routing
3.2.1 Least Cost Routing - An implementation example
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 Intelligent vs. simple terminals Signalling Inter-department trunking Legacy functionality Wireless VoIP Issues
3.4 Scenario 3: Integration of VoIP and Videoconferencing
3.4.1 Integrating Voice and Videoconferencing over IP - an example
Chapter 4
Setting up basic services
4.1 General concepts
4.1.1 Architecture PSTN gateways / PBX migration
4.1.2 Robustness
4.1.3 Management issues Multiple account databases Decentralization
4.2 Dialplans
4.3 Authentication
4.3.1 Authentication in H.323 Areas of application User Authentication Integrity Confidentiality Security profiles Basic Scenarios
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) Installation Configuration Operation Endpoint authentication Advanced features
4.5.2 Using a Radvision Enhanced Communication Server (ECS gatekeeper) Installation Configuration Operation Endpoint authentication Advanced features
4.5.3 Using an OpenH323 Gatekeeper - GNU Gatekeeper Installation Configuration Operation Endpoint authentication Advanced features
4.6 Setting up SIP services
4.6.1 Operation of SIP Servers Recommended Operational Practices
4.6.2 SIP Express Router Getting SIP Express Router Installation (From binary packages) MySQL setup Configuration Operation
4.6.3 Asterisk Getting Asterisk Installation Configuration
4.6.4 VOCAL Overview Installation Configuration Operation Endpoint authentication Advanced Features
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 Overview Support in SIP User Agents Support in SIP Server RTP Proxy Real World Setup
Chapter 5
Setting up Advanced Services
5.1 Gatewaying
5.1.1 Gateway interfaces Subscriber Loop E&M interfaces E1/CAS trunk ISDN Access Interfaces
5.1.2 Gatewaying from H.323 to PSTN/ISDN Using a RADVISION OnLAN 323 L2W-323 Gateway sip-ua Configuration
5.1.4 Gatewaying from SIP to H.323 and vice versa User Registration Call from SIP to H.323 Call from H.323 to SIP Media Switching and Capability Negotiation Call termination Configuration guidelines
5.1.5 Accounting Gateways
5.2 Supplementary services
5.2.1 Supplementary Services using H.323 Call Transfer Supplementary Service Call Diversion Supplementary Service Call Waiting Supplementary Service Supplementary services (H.450) support in popular gatekeepers
5.2.2 Supplementary Services using SIP On Hold Call Transfer Unconditional Call Forwarding Conditional Forwarding
5.3 Multipoint Conferencing
Chapter 6
Setting up Value-Added Services
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 Installation Configuration Operation
6.2.5 SIP Express Router Message Store
6.3 Voicemail
Chapter 7
Global telephony integration
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) Structure Addressing Protocol
7.1.4 SRV-Records
7.1.5 ENUM
7.2 Call routing today
7.2.1 SIP
7.2.2 Using H.323 Global Dialing Scheme Problems
7.3 Utopia: Setting up global IP telephony
7.4 Towards Utopia
7.4.1 Call Routing Assistant
Chapter 8
Regulatory / Legal considerations
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 New Regulatory Framework - an overview
8.3.4 Authorization System instead of Licensing System Example: VoIP in the New Framework in the UK
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
European IP Telephony Projects
A.1 Evolute
A.2 6Net
A.3 Eurescom P1111 (Next-Gen open Service Solutions over IP (N- GOSSIP)
A.5 The GRNET/RTS project
A.6 SURFWorks
A.7 VC Stroom
A.8 Voice services in the CESNET2 network
B.1 Softphones
B.2 Hardphones
B.3 Servers
B.4 Gateways
B.5 Testing
B.6 Miscellaneous
Media Gateway Control Protocol ( MGCP )
Public Switched Telephone Network ( MGCP )
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Voip Cookbook

Voip Cookbook

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Published by qillisse

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Published by: qillisse on Jan 15, 2011
Copyright:Attribution Non-commercial


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