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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
1.5. Techno-economic aspect of moving from classic telephony to VoIP
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
Figure 2.1. Scope and Components defined in H.323
Figure 2.2. H.323 protocol architecture
2.2.1.4. Signaling models
Figure 2.5. Gatekeeper Routed call signaling model
Figure 2.6. Gatekeeper Routed H.245 control model
Figure 2.7. OPENLOGICALCHANNELACK message content
2.2.1.6. Locating zone external targets
Locating zone external targets
2.2.2. SIP
Figure 2.13. Registrar Overview
2.2.2.4. SIP Transactions
Figure 2.18. REGISTER Message Flow
Figure 2.20. BYE Message Flow (With and without Record Routing)
2.2.2.6.5. Event Subscription And Notification
Event Subscription And Notific- ation
2.2.2.6.6. Instant Messages
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)
Proprietary Signaling Protocols
Chapter 3. IP Telephony Scenarios
3.1. Introduction
3.2. Scenario 1: Long-distance least cost routing
Figure 3.1. Traditional separation of data and telephony between locations
Figure 3.3. Least cost routing architecture
3.2.1. Least Cost Routing - An example of integration
3.3. Scenario 2: Legacy PBX replacement
Figure 3.4. Legacy PBX which trunks to the PSTN
3.3.1. Scenario 2a: IP-Phone without PBX
Scenario 2: Legacy PBX re- placement
Figure 3.5. IP-Phone to IP-Phone without PBX
3.3.2. Scenario 2b: Integration with legacy PBX sys- tems
Scenario 2b: Integration with legacy PBX systems
3.3.3. Scenario 2c: Full replacement
Scenario 2c: Full replacement
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
Figure 4.2. SIP/H.323 zone using a signaling gateway
Figure 4.3. Routing based on number prefix
Figure 4.5. Per number routing with a) two or b) one gateways
Figure 4.6. Prefix based trunking
Figure 4.8. Dynamic individual trunking
4.1.2. Robustness
4.1.3. Management issues
4.1.4. Operation of SIP Servers
Example 4.2. Use of SIPSak for Learning SIP Path
4.2. Dialplans
4.3. Authentication and Billing
4.3.1. Authentication in H.323
Authentication and Billing
4.3.2. Authentication in SIP
Figure 4.9. REGISTER Message Flow
4.4. Examples
4.4.1. Example 1: Simple, use IP telephony like legacy telephony
4.4.2. Example 2: Complex, full featured
Figure 4.12. Example of a multi-server IP telephony zone
4.5. Setting up H.323 services
Setting up H.323 services
Figure 4.13. Gatekeeper features examples
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 OpenH323 Gatekeeper - GNU Gatekeep- er
4.6. Setting up SIP services
4.6.1. SIP Express Router
4.6.1.4.2. Default Configuration Script
Example 4.3. Default Configuration Script
Example 4.6. Configuration with Enabled Accounting
Example 4.7. Configuration of Use of Aliases
Example 4.8. Script for Gateway Access Control
4.6.2. Asterisk
4.6.3. 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
5.1. Gatewaying
5.1.1. Gateway interfaces
Figure 5.1. The role of PBX to voice gateway interface
Figure 5.3. ISDN configuration
Figure 5.4. Q.931 call control messages in call-setup with the en-bloc signal
5.1.2. Gatewaying from H.323 to PSTN/ISDN
Figure 5.6. CISCO voice gateway interconnection
5.1.3. Gatewaying from SIP to PSTN/ISDN
5.1.4. Gatewaying from SIP to H.323 and vice versa
Figure 5.7. SIP / H.323 gateway containing SIP proxy and registrar
Figure 5.8. SIP / H.323 gateway containing a H.323 gatekeeper
5.1.5. Accounting Gateways
5.2. Supplementary services
5.2.1. Supplementary Services using H.323
Figure 5.10. Messages exchanged to implement the CT-SS without Gatekeeper
5.2.2. Supplementary Services using SIP
Supplementary Services using SIP
Figure 5.13. On-hold Call Flow
5.3. Multipoint Conferencing
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
Figure 6.1. REFER Based Click-to-Dial
6.2.2. Presence
6.2.3. Missed Calls
6.2.4. Serweb
Figure 6.2. Serweb - My Account
Figure 6.3. Serweb - Phonebook
Figure 6.4. Serweb - Missed Calls
Figure 6.5. Serweb - Message Store
6.2.5. SIP Express Router Message Store
6.3. Voicemail
7.1. Technology
7.1.1. H.323 LRQ
7.1.2. H.225.0 Annex G
7.1.3. TRIP
7.1.4. SRV-Records
7.1.5. ENUM
7.2. Today
7.2.1. H.323
7.2.2. SIP
7.3. Migration
7.4. The future
8.8. Unbundling
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MainIP Telephony CookBook

MainIP Telephony CookBook

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Published by Ilyas Sayyad

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Published by: Ilyas Sayyad on Sep 05, 2011
Copyright:Attribution Non-commercial

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12/15/2012

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