Name: Carrier Grade Voice Over IP von Daniel Collins
SKU: 9780071406345
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Carrier Grade Voice Over IP Zusammenfassung

Carrier Grade Voice Over IP Daniel Collins

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.

Über Daniel Collins

Daniel Collins played key roles in the development of 2G wireless systems and the introduction of GSM to North America during nearly a decade of work with Ericsson in the United States and the United Kingdom. Now an independent consultant, he specializes in VoIP and 3G wireless network architectures.

Inhaltsverzeichnis

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

Zusätzliche Informationen

Sku

GOR005391895

ISBN 13

9780071406345

ISBN 10

0071406344

Titel

Carrier Grade Voice Over IP Daniel Collins

Autor

Daniel Collins

Zustand

Gebraucht - Sehr Gut

Bindung

Broschiert

Verlag

McGraw-Hill Education - Europe

Datum

20021016

Anzahl der Seiten

522

Preise

N/A

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