Carrier-grade VoIP / Richard Swale, Daniel Collins.

Previous edition by Daniel Collins; published 2003.

Includes bibliographical references (pages 471-480) and index.

Machine generated contents note: 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 -- Alternative 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 -- Endnotes -- 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 -- The Internet Engineering Task Force -- The Internet Engineering Steering Group -- The Internet Assigned Numbers Authority -- The Internet Standards Process -- The Internet Protocol -- The IP Header -- IP Routing -- IP Version 6 -- IPv6 Header -- IPv6 Addresses -- IPv6 Header Extensions -- Interworking IPv4 and IPv6 -- IP Multicast -- The Transmission Control Protocol -- The TCP Header -- TCP Connections -- The User Datagram Protocol -- Voice over UDP, Not TCP -- The Stream Control Transmission Protocol -- Some SCTP Concepts -- Types of SCTP Chunks -- Establishing an Association -- Transferring Data -- The Real-Time Transport Protocol -- RTP Payload Formats -- The RTP Header -- The RTP Control Protocol -- RTCP Sender Report -- RTCP Receiver Report -- RTCP Source Description Packet -- RTCP Bye Packet -- Application-Defined RTCP Packet -- Timing of RTCP Packets -- Calculating Round-Trip Time -- Calculating Jitter -- Security and Performance Optimization -- The Secure Real-Time Transport Protocol -- Header Compression -- Endnotes -- 3.Speech-Coding Techniques -- A Little about Speech, Audio, and Music -- Exploring the Boundaries -- Voice Sampling -- Quantization -- Framing and Packetization -- Wideband, Stereo and HD Voice -- Voice Quality -- The E-Model -- Testing Methods -- Types of Speech Coders -- Waveform Coders -- G.711.0 -- G.711.1 -- Adaptive Differential PCM -- Analysis-by-Synthesis Codecs -- G.728 Low-Delay CELP -- G.723.1 -- G.729 Conjugate-Structure Algebraic CELP (CS-Acelp) -- G.729.1 -- G.722-Wideband Audio -- G.722.1 -- G.722.2 -- Codecs for the Internet -- The Internet Low Bit Rate Codec: A Narrowband Codec for the Internet -- Opus: A Wideband Codec for the Internet -- Other Codecs -- Transcoding Speech -- Mixers and Translators -- Tones, Signals, and Dual-Tone Multifrequency Digits -- Choosing a Codec -- Endnotes -- 4.H.323: Multimedia Conferencing over IP -- The H.323 Architecture -- H.323 Gatekeeper -- H.323 Multipoint Control Unit -- H.323 Gateways -- Overview of H.323 Signaling -- 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 -- Fast Everything -- Conference Calls -- Prearranged Conference -- Ad Hoc Conference -- Securing an H.323 Network -- Future Directions -- Endnotes -- 5.The Session Initiation Protocol -- The Popularity of SIP -- The SIP Architecture -- SIP Network Entities -- SIP Call Establishment -- Advantages of SIP 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 Servers -- Proxy Servers -- Back-to-Back User Agent -- The Session Description Protocol -- The Structure of SDP -- SDP Syntax -- Usage of SDP with SIP -- Negotiation of Media -- SIP Extensions and Enhancements -- 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 -- Building Call-Handling Applications -- Interworking -- PSTN Interworking -- Interworking with H.323 -- Interworking with Web Browsers Using WebRTC -- SIP Futures -- Summary -- Endnotes -- 6.Distributed Gateways and the Softswitch Architecture -- Separation of Media and Call Control -- Softswitch Architecture -- Protocol Requirements for Controlling Media Gateways -- Protocols for Controlling Media Gateways -- 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 -- Megacop/H.248.1 -- H.248.1 Architecture -- Overview of H.248.1 Commands -- Descriptors -- Packages -- H.248.1 Command and Response Details -- Call Setup Using H.248.1 -- Interworking between H.248.1 and SIP -- Future Direction of H.248.1 -- Endnote -- 7.VoIP and SS7 -- The SS7 Protocol Suite -- The Message Transfer Part -- ISDN User Part, Signaling Connection Control Part, and Transaction Capabilities -- SS7 Network Architecture -- Signaling Points -- Signal Transfer Point -- Service Control Point -- Message Signal Units -- SS7 Addressing -- ISUP -- Performance Requirements for SS7 -- Sigtran -- Sigtran Architecture -- SCTP -- M3UA Operation -- M2UA Operation -- M2PA Operation -- Interworking SS7 and VoIP Architectures -- Interworking Softswitch and SS7 -- Interworking H.323 and SS7 -- Endnotes -- 8.Quality of Service -- 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 Syntax -- Establishing Reservations -- Reservation Errors -- Guaranteed Service -- Controlled-Load Service -- Removing Reservations and the Use of Soft State -- Changing Existing Reservations -- DiffServ -- The DiffServ Architecture -- The Need for SLAs -- PHB -- Multiprotocol Label Switching -- The MPLS Architecture -- FEC and Label Formats -- Actions at LSRs -- MPLS Traffic Engineering -- Label Distribution Protocols and Constraint-Based Routing -- RSVP Traffic Engineering -- Combining QoS Solutions -- Endnotes -- 9.Interconnecting VoIP Networks -- Introduction to Middleboxes: The Problem with Connecting VoIP Networks -- VoIP Is a Session-Based Application -- An Example: Differences in Address Format and Policy -- Impact of Middleboxes on VoIP Networks -- Common Problems When Interconnecting VoIP Networks -- The Firewall Problem -- The Addressing Problem -- The QoS Problem -- The Border Gateway Problem -- The Security Authentication and Encryption Problem -- Overview of VoIP Interconnect Solutions -- Middlebox Communications -- Border Elements and Session Border Controllers -- Achieving Carrier-Grade VoIP Interconnection -- Network Security Policy -- The Vulnerabilities of VoIP Systems -- Why Can VoIP Systems Be Insecure? -- Tools for Addressing VoIP Security -- Interconnecting Carrier-Grade VoIP Networks -- User-to-Network Interface -- IP Trunking -- SIP Trunking -- Network-to-Network Interface -- VoIP Address Resolution -- Bilateral Agreement Model -- Clearinghouse Model -- Endnotes -- 10.Designing a Voice over IP Network -- Design Criteria -- Build-Ahead or Capacity Buffer -- Fundamental Technology Assumptions -- Network-Level Redundancy -- Voice Coder/Decoder 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 Usage Forecast -- Traffic Distribution Forecast -- Node Locations and Bandwidth Requirements -- MG Locations and PSTN Trunk Dimensioning -- MGC, SG, and EMS Dimensioning and

Placement -- Calculating VoIP Bandwidth Requirements -- Physical Connectivity -- Further Exercises -- IPv6 -- Adding SIP Trunks -- Multiple Codec Support -- Endnotes.