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Table Of Contents

Introduction
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
2.2.1.1 Scope
2.2.1.2 Signaling protocols
2.2.1.3 Gatekeeper Discovery and Registration
2.2.1.4 Signaling models
2.2.1.5 Communication Phases
2.2.1.6 Locating zone external targets
2.2.1.7 Sample Call Scenario
2.2.1.8 Additional (Call) Services
2.2.1.9 H.235 Security
2.2.1.10 Protocol Profiles
2.2.2 SIP
2.2.2.1 Purpose of SIP
2.2.2.2 SIP Network Elements
2.2.2.3 SIP Messages
2.2.2.4 SIP Transactions
2.2.2.5 SIP Dialogs
2.2.2.6 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)
2.2.5.1 RTP Header
2.2.5.2 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
3.3.3.1 Intelligent vs. simple terminals
3.3.3.2 Signalling
3.3.3.3 Inter-department trunking
3.3.3.4 Legacy functionality
3.3.3.5 Wireless VoIP
3.3.3.6 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
4.1.1.1 PSTN gateways / PBX migration
4.1.2 Robustness
4.1.3 Management issues
4.1.3.1 Multiple account databases
4.1.3.2 Decentralization
4.2 Dialplans
4.3 Authentication
4.3.1 Authentication in H.323
4.3.1.1 Areas of application
4.3.1.2 User Authentication
4.3.1.3 Integrity
4.3.1.4 Confidentiality
4.3.1.5 Security profiles
4.3.2.3 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)
4.5.1.1 Installation
4.5.1.2 Configuration
4.5.1.3 Operation
4.5.1.4 Endpoint authentication
4.5.1.5 Advanced features
4.5.2 Using a Radvision Enhanced Communication Server (ECS gatekeeper)
4.5.2.1 Installation
4.5.2.2 Configuration
4.5.2.3 Operation
4.5.2.4 Endpoint authentication
4.5.2.5 Advanced features
4.5.3 Using an OpenH323 Gatekeeper - GNU Gatekeeper
4.5.3.1 Installation
4.5.3.2 Configuration
4.5.3.3 Operation
4.5.3.4 Endpoint authentication
4.5.3.5 Advanced features
4.6 Setting up SIP services
4.6.1 Operation of SIP Servers
4.6.1.1 Recommended Operational Practices
4.6.2 SIP Express Router
4.6.2.1 Getting SIP Express Router
4.6.2.2 Installation (From binary packages)
4.6.2.3 MySQL setup
4.6.2.4 Configuration
4.6.2.5 Operation
4.6.3 Asterisk
4.6.3.1 Getting Asterisk
4.6.3.2 Installation
4.6.3.3 Configuration
4.6.4 VOCAL
4.6.4.1 Overview
4.6.4.2 Installation
4.6.4.3 Configuration
4.6.4.4 Operation
4.6.4.5 Endpoint authentication
4.6.4.6 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
4.7.3.1 Overview
4.7.3.2 Support in SIP User Agents
4.7.3.3 Support in SIP Server
4.7.3.4 RTP Proxy
4.7.3.5 Real World Setup
Chapter 5
Setting up Advanced Services
5.1 Gatewaying
5.1.1 Gateway interfaces
5.1.1.1 Subscriber Loop
5.1.1.2 E&M interfaces
5.1.1.3 E1/CAS trunk
5.1.1.4 ISDN Access Interfaces
5.1.2 Gatewaying from H.323 to PSTN/ISDN
5.1.2.1 Using a RADVISION OnLAN 323 L2W-323 Gateway
5.1.3.2 sip-ua Configuration
5.1.4 Gatewaying from SIP to H.323 and vice versa
5.1.4.1 User Registration
5.1.4.2 Call from SIP to H.323
5.1.4.3 Call from H.323 to SIP
5.1.4.4 Media Switching and Capability Negotiation
5.1.4.5 Call termination
5.1.4.6 Configuration guidelines
5.1.5 Accounting Gateways
5.2 Supplementary services
5.2.1 Supplementary Services using H.323
5.2.1.1 Call Transfer Supplementary Service
5.2.1.2 Call Diversion Supplementary Service
5.2.1.3 Call Waiting Supplementary Service
5.2.1.4 Supplementary services (H.450) support in popular gatekeepers
5.2.2 Supplementary Services using SIP
5.2.2.1 On Hold
5.2.2.2 Call Transfer
5.2.2.3 Unconditional Call Forwarding
5.2.2.4 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
6.2.4.1 Installation
6.2.4.2 Configuration
6.2.4.3 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)
7.1.3.1 Structure
7.1.3.2 Addressing
7.1.3.3 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
7.2.2.1 Global Dialing Scheme
7.2.2.2 Problems
7.3. UTOPIA: SETTING UP GLOBAL IP TELEPHONY
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
8.3.4.1 Example: VoIP in the New Framework in the UK
8.3.5 Numbering
8.3.6 Access
8.3.7 Interconnection
8.4. VOICE OVER IP IN THE UNITED STATES
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.4 HITEC
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
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05/23/2012

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