Carrier Grade Voice Over IP

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October 8, 2002


Electronic book text, 522 pages


0071501118 / 9780071501118

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Main description

In 2002 voice over IP will constitute more than 25% of all long distance voice calls, according to Network World. That's more than a 30% ramp-up from 2001. The emergence of SIP, MPLS and new quality of service tools is making carrier grade voice over IP a service reality, and a potentially huge margin booster and revenue driver for service providers. The first edition of Carrier Grade Voice over IP played a roll in VoIP growth, in less than year becoming an essential tool for carriers working to provide high quality IP telephony. This new edition vastly updates the SIP chapter, details MPLS, and takes the explanations of the previous edition a step further in a final chapter that shows, step by step, how to design working VoIP networks.

Table of contents

CHAPTER 1: INTRODUCTION What is Meant by Carrier-Grade? What is Meant by VoIP? A Little About IP Why VoIP? Why Carry Voice? Why Use IP for Voice? Lower Equipment Cost Voice/Data Integration and Advanced Services Potentially Lower Bandwidth Requirements The Widespread Availability of IP The VoIP Market VoIP Challenges Speech Quality Managing Access and Prioritizing Traffic Speech-Coding Techniques Network Reliability and Scalability Overview of the Following Chapters CHAPTER 2: TRANSPORTING VOICE BY USING IP Overview of the IP Protocol Suite Internet Standards and the Standards Process The Internet Society The Internet Architecture Board (IAB) The Internet Engineering Task Force (ETF) The Internet Engineering Steering Group (ESG) The Internet Assigned Numbers Authority (IANA) The Internet Standards Process The Internet Prototol (IP) The IP Header IP Routing The Transmission Control Protocol (TCP) The TCP Header TCP Connections The User Datagram Protocol (UDP) Voice over UDP, not TCP The Real-Time Transport Protocol (RTP) RTP Payload Formats The RTP Header Mixers and Translators The RTP Control Protocol (RTCP) RTCP Sender Report (SR) RTCP Receiver Report (RR) RTCP Source Description Packet (SDES) RTCP BYE Packet Application-Defined RTCP Packet Calculating Round-Trip Time Calculating Jitter Timing of RTCP Packets IP Multicast IP Version 6 (IPv6) IPv6 Header IPv6 Addresses IPv6 Header Extensions Interworking IPv4 and IPv6 CHAPTER 3: SPEECH-CODING TECHNIQUES Voice Quality A Little About Speech Voice Sampling Quantization Types of Speech Coders G.711 Adaptive Differential PCM (ADPCM) Analysis-by-Synthesis (AbS) Codecs G.728 Low-Delay CELP (LD-CELP) G.723.1 Algebraic Code-Excited Linear Prediction (ACELP) G.729 Selecting Codecs Cascaded Codecs Tones, Signals, and Dual-Tone Multifrequency (DTMF) Digits CHAPTER 4: H.323 The H.323 Architecture Overview of H.323 Signaling Overview of H.323 Protocols H.323 Addressing Codecs RAS Signaling Gatekeeper Discovery Endpoint Registration and Registration Cancellation Endpoint Location Admission Bandwidth Change Status Disengage Resource Availability Service Control Request in Progress Call Signaling Setup Call-Proceeding Alerting Progress Connect Release Complete Facility Interaction Between Call Signaling and H.245 Control Signaling Call Scenarios Basic Call Without Gatekeepers A Basic Call with Gatekeepers and Direct Endpoint Call Signaling A Basic Call with Gatekeeper/Direct Routed Call Signaling A Basic Call with Gatekeeper-Routed Call Signaling Optional Called-Endpoint Signaling H.245 Control Signaling H.245 Message Groupings The Concept of Logical Channels H.245 Procedures Fast Connect Procedure H.245 Message Encapsulation Conference Calls Pre-arranged Conference Ad Hoc Conference The Decomposed Gateway CHAPTER 5: THE SESSION INITIATION PROTOCOL (SIP) The Popularity of SIP The SIP Architecture SIP Network Entities SIP Call Establishment SIP Advantages over Other Signaling Protocols Overview of SIP Messaging Syntax SIP Requests SIP Responses SIP Addressing Message Headers Examples of SIP Message Sequences Registration Invitation Termination of a Call Redirect and Proxy Servers Redirect Services Proxy Servers The Session Description Protocol (SDP) The Structure of SDP SDP Syntax Usage of SDP with SIP Negotiation of Media SIP Extensions and Enhancements The SIP INFO Method SIP Event Notification SIP for Instant Messaging The SIP REFER Method Reliability of Provisional Responses The SIP UPDATE Method Integration of SIP Signaling and Resource Management Usage of SIP for Features and Services Call Forwarding Consultation Hold Interworking PSTN Interworking Interworking with H.323 Summary CHAPTER 6: MEDIA GATEWAY CONTROL AND THE SOFTSWITCH ARCHITECTURE Separation of Media and Call Control Softswitch Architecture Requirements for Media Gateway Control Protocols for Media Gateway Control MGCP The MGCP Model MGCP Endpoints MGCP Calls and Connections Overview of MGCP Commands Overview of MGCP Responses Command and Response Details Call Setup Using MGCP MGCP Events, Signals, and Packages Interworking Between MGCP and SIP MEGACO.248 MEGACO Architecture Overview of MEGACO Commands Descriptors Packages MEGACO Command and Response Details Call Setup Using MEGACO Interworking Between MEGACO and SIP CHAPTER 7: VoIP and SS7 The SS7 Protocol Suite The Message Transfer Part (MTP) ISDN User Part (ISUP) and Signaling Connection Control Part (SCCP) SS7 Network Architecture Signaling Points (SPs) Signal Transfer Point (STP) Service Control Point (SCP) Message Signal Units (MSUs) SS7 Addressing ISUP Performance Requirements for SS& Sigtran Sigtran Architecture SCTP M3UA Operation M2UA Operation M2PA Operation Interworking SS7 and VoIP Architectures Interworking Softswitch and SS7 Interworking H.323 and SS7 CHAPTER 8: QUALITY OF SERVICE (QoS) The Need for QoS End-to-End QoS It's Not Just the Network Overview of QoS Solutions More Bandwidth QoS Protocols and Architectures QoS Policies The Resource Reservation Protocol (RSVP) RSVP Syntax Establishing Reservations Reservation Errors Guaranteed Service Controlled-Load Service Removing Reservations and the Use of Soft State DiffServ The DiffServ Architecture The Need for SLAs Per-Hop Behavior (PHB) Multiprotocol Label Switching (MPLS) The MPLS Architecture FEC and Label Formats Actions at LSRs MPLS Traffic Engineering Label Distribution Protocols and Constraint-Based Routing RSVP Traffic Engineering (RSVP-TE) Combining QoS Solutions CHAPTER 9: DESIGNING A VOICE OVER IP NETWORK Design Criteria Build-Ahead or Capacity Buffer Fundamental Technology Assumptions Network-Level Redundancy Voice Coder/Decoder (Codec) Selection Issues Blocking Probability QoS Protocol Considerations and Layer 2 Protocol Choices Product and Vendor Selection Generic VoIP Product Requirements Element Management Traffic Forecasts Voice Usae Forecast Traffic Distribution Forecast Node Locations and Bandwidth Requirements MG Locations and PSTN Trunk Dimensioning MSG, SG, and EMS Dimensioning and Placement Calculating VoIP Bandwidth Requirements Physical Connectivity APPENDIX A: TABLE OF ERLANG B APPENDIX B: VISUAL BASIC CODE FOR ERLANG CALCULATIONS Glossary of Acronyms References Index

Author comments

Daniel Collins has worked in the telecommunications industry for 15 years. He spent approximately nine years with Ericsson in various countries, including Ireland, Australia, the United Kingdom and the United States. During that time he worked extensively with both wireline and wireless network technologies. He helped to develop and deploy 2G wireless systems in Europe; he played a major role in the adaptation of GSM standards for use in the United States; and he was a major contributor to the launch of some of the earliest PCS networks in North America.

Since leaving Ericsson, Daniel has worked for a new telecommunications carrier and, more recently as a consultant. In a consultancy capacity, he has provided wireless and VoIP engineering expertise to numerous network operators, consultancy companies and infrastructure vendors. Daniel's clients include PrimeCo Personal Communications (now part of Verizon Wireless), Synacom Technology, AT&T Wireless, Alcatel USA and several other companies.

Collins is co-author of 3G Wireless Networks, published by McGraw-Hill. He holds a degree in Electrical and Electronic engineering from the National University of Ireland.

Back cover copy


In the race to put carrier-quality voice over IP (VoIP)----those using this book run faster and falter less. International expert Daniel Collins' greatly enhanced Carrier Grade Voice over IP brings you leading-edge signaling schemes, protocol apps, and quality of service (QoS) techniques--and more importantly, the answers to making them work together.

Skipping needless history, chitchat, and math, Collins gets right down to solutions with solid information on protocols' purposes and uses. He gives you the newest information with hands-on details, showing you how to:

* Use ITU and IETF protocols to build next-generation networks
* Enable advanced features and services, including expanded SIP coverage
* Provide wireline-quality service with resource reservation schemes and QoS techniques, including the latest information on MPLS advances
* Overcome limitations inherent in IP
* Seamlessly interwork VoIP systems with traditional telephony networks
* Design real-world VoIP networks (new chapter)
* Pull the pieces together into a system that works--and does not break

Complete with numerous specific examples of how the protocols are used and integrated, this just-in-time guide supplies the solutions you need to roll out competitive quality VoIP.

Jumpstart carrier-quality VoIP with help from international expert Daniel Collins

Copyright 2014 McGraw-Hill Global Education Holdings, LLC


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