Carrier Grade Voice Over IP

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Date

September 17, 2002

Format

Electronic book text, 522 pages

ISBN

0071501118 / 9780071501118

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65.00



Overview


Main description

NEW EDITION OF---CUTTING-EDGE SOLUTIONS CUT-TO-THE-CHASE STYLE

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


Table of contents

CHAPTER 1: INTRODUCTIONWhat is Meant by Carrier-Grade?What is Meant by VoIP?A Little About IPWhy VoIP?Why Carry Voice?Why Use IP for Voice?Lower Equipment CostVoice/Data Integration and Advanced ServicesPotentially Lower Bandwidth RequirementsThe Widespread Availability of IPThe VoIP MarketVoIP ChallengesSpeech QualityManaging Access and Prioritizing TrafficSpeech-Coding TechniquesNetwork Reliability and ScalabilityOverview of the Following ChaptersCHAPTER 2: TRANSPORTING VOICE BY USING IPOverview of the IP Protocol SuiteInternet Standards and the Standards ProcessThe Internet SocietyThe 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 ProcessThe Internet Prototol (IP)The IP HeaderIP RoutingThe Transmission Control Protocol (TCP)The TCP HeaderTCP ConnectionsThe User Datagram Protocol (UDP)Voice over UDP, not TCPThe Real-Time Transport Protocol (RTP)RTP Payload FormatsThe RTP HeaderMixers and TranslatorsThe RTP Control Protocol (RTCP)RTCP Sender Report (SR)RTCP Receiver Report (RR)RTCP Source Description Packet (SDES)RTCP BYE PacketApplication-Defined RTCP PacketCalculating Round-Trip TimeCalculating JitterTiming of RTCP PacketsIP MulticastIP Version 6 (IPv6)IPv6 HeaderIPv6 AddressesIPv6 Header ExtensionsInterworking IPv4 and IPv6CHAPTER 3: SPEECH-CODING TECHNIQUESVoice QualityA Little About SpeechVoice SamplingQuantizationTypes of Speech CodersG.711Adaptive Differential PCM (ADPCM)Analysis-by-Synthesis (AbS) CodecsG.728 Low-Delay CELP (LD-CELP)G.723.1 Algebraic Code-Excited Linear Prediction (ACELP)G.729Selecting CodecsCascaded CodecsTones, Signals, and Dual-Tone Multifrequency (DTMF) DigitsCHAPTER 4: H.323The H.323 ArchitectureOverview of H.323 SignalingOverview of H.323 ProtocolsH.323 AddressingCodecsRAS SignalingGatekeeper DiscoveryEndpoint Registration and Registration CancellationEndpoint LocationAdmissionBandwidth ChangeStatusDisengageResource AvailabilityService ControlRequest in ProgressCall SignalingSetupCall-ProceedingAlertingProgressConnectRelease CompleteFacilityInteraction Between Call Signaling and H.245 Control SignalingCall ScenariosBasic Call Without GatekeepersA Basic Call with Gatekeepers and Direct Endpoint Call SignalingA Basic Call with Gatekeeper/Direct Routed Call SignalingA Basic Call with Gatekeeper-Routed Call SignalingOptional Called-Endpoint SignalingH.245 Control SignalingH.245 Message GroupingsThe Concept of Logical ChannelsH.245 ProceduresFast Connect ProcedureH.245 Message EncapsulationConference CallsPre-arranged ConferenceAd Hoc ConferenceThe Decomposed GatewayCHAPTER 5: THE SESSION INITIATION PROTOCOL (SIP)The Popularity of SIPThe SIP ArchitectureSIP Network EntitiesSIP Call EstablishmentSIP Advantages over Other Signaling ProtocolsOverview of SIP Messaging SyntaxSIP RequestsSIP ResponsesSIP AddressingMessage HeadersExamples of SIP Message SequencesRegistrationInvitationTermination of a CallRedirect and Proxy ServersRedirect ServicesProxy ServersThe Session Description Protocol (SDP)The Structure of SDPSDP SyntaxUsage of SDP with SIPNegotiation of MediaSIP Extensions and EnhancementsThe SIP INFO MethodSIP Event NotificationSIP for Instant MessagingThe SIP REFER MethodReliability of Provisional ResponsesThe SIP UPDATE MethodIntegration of SIP Signaling and Resource ManagementUsage of SIP for Features and ServicesCall ForwardingConsultation HoldInterworkingPSTN InterworkingInterworking with H.323SummaryCHAPTER 6: MEDIA GATEWAY CONTROL AND THE SOFTSWITCH ARCHITECTURESeparation of Media and Call ControlSoftswitch ArchitectureRequirements for Media Gateway ControlProtocols for Media Gateway ControlMGCPThe MGCP ModelMGCP EndpointsMGCP Calls and ConnectionsOverview of MGCP CommandsOverview of MGCP ResponsesCommand and Response DetailsCall Setup Using MGCPMGCP Events, Signals, and PackagesInterworking Between MGCP and SIPMEGACO.248MEGACO ArchitectureOverview of MEGACO CommandsDescriptorsPackagesMEGACO Command and Response DetailsCall Setup Using MEGACOInterworking Between MEGACO and SIPCHAPTER 7: VoIP and SS7The SS7 Protocol SuiteThe Message Transfer Part (MTP)ISDN User Part (ISUP) and Signaling Connection Control Part (SCCP)SS7 Network ArchitectureSignaling Points (SPs)Signal Transfer Point (STP)Service Control Point (SCP)Message Signal Units (MSUs)SS7 AddressingISUPPerformance Requirements for SS&SigtranSigtran ArchitectureSCTPM3UA OperationM2UA OperationM2PA OperationInterworking SS7 and VoIP ArchitecturesInterworking Softswitch and SS7Interworking H.323 and SS7CHAPTER 8: QUALITY OF SERVICE (QoS)The Need for QoSEnd-to-End QoSIt's Not Just the NetworkOverview of QoS SolutionsMore BandwidthQoS Protocols and ArchitecturesQoS PoliciesThe Resource Reservation Protocol (RSVP)RSVP SyntaxEstablishing ReservationsReservation ErrorsGuaranteed ServiceControlled-Load ServiceRemoving Reservations and the Use of Soft StateDiffServThe DiffServ ArchitectureThe Need for SLAsPer-Hop Behavior (PHB)Multiprotocol Label Switching (MPLS)The MPLS ArchitectureFEC and Label FormatsActions at LSRsMPLS Traffic EngineeringLabel Distribution Protocols and Constraint-Based RoutingRSVP Traffic Engineering (RSVP-TE)Combining QoS SolutionsCHAPTER 9: DESIGNING A VOICE OVER IP NETWORKDesign CriteriaBuild-Ahead or Capacity BufferFundamental Technology AssumptionsNetwork-Level RedundancyVoice Coder/Decoder (Codec) Selection IssuesBlocking Probability QoS Protocol Considerations and Layer 2 Protocol ChoicesProduct and Vendor SelectionGeneric VoIP Product RequirementsElement ManagementTraffic ForecastsVoice Usae ForecastTraffic Distribution ForecastNode Locations and Bandwidth RequirementsMG Locations and PSTN Trunk DimensioningMSG, SG, and EMS Dimensioning and PlacementCalculating VoIP Bandwidth RequirementsPhysical ConnectivityAPPENDIX A: TABLE OF ERLANG BAPPENDIX B: VISUAL BASIC CODE FOR ERLANG CALCULATIONSGlossary of AcronymsReferencesIndex


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

NEW EDITION OF---CUTTING-EDGE SOLUTIONS CUT-TO-THE-CHASE STYLE

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





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