IP Telephony E-Book

Olivier Hersent

• Herausgeber: John Wiley & Sons
• Kategorie: Wissenschaft und neue Technologien
• Sprache: Englisch

Seitenzahl: 745

Veröffentlichungsjahr: 2011

Beschreibung

All you need to know about deploying VoIP protocols in one comprehensive and highly practical reference - Now updated with coverage on SIP and the IMS infrastructure This book provides a comprehensive and practical overview of the technology behind Internet Telephony (IP), providing essential information to Network Engineers, Designers, and Managers who need to understand the protocols. Furthermore, the author explores the issues involved in the migration of existing telephony infrastructure to an IP - based real time communication service. Assuming a working knowledge of IP and networking, it addresses the technical aspects of real-time applications over IP. Drawing on his extensive research and practical development experience in VoIP from its earliest stages, the author provides an accessible reference to all the relevant standards and cutting-edge techniques in a single resource. Key Features: * Updated with a chapter on SIP and the IMS infrastructure * Covers ALL the major VoIP protocols - SIP, H323 and MGCP * Includes a large section on practical deployment issues gleaned from the authors' own experience * Chapter on the rationale for IP telephony and description of the technical and business drivers for transitioning to all IP networks This book will be a valuable guide for professional network engineers, designers and managers, decision makers and project managers overseeing VoIP implementations, market analysts, and consultants. Advanced undergraduate and graduate students undertaking data/voice/multimedia communications courses will also find this book of interest. Olivier Hersent founded NetCentrex, a leading provider of VoIP infrastructure for service providers, then became CTO of Comverse after the acquisition of NetCentrex. He now manages Actility, provider of IMS based M2M and smartgrid infrastructure and applications.

Leseprobe

Contents

Abbreviations

Glossary

Preface

1 Multimedia Over Packet

1.1 TRANSPORTING VOICE, FAX, AND VIDEO OVER A PACKET NETWORK

1.2 ENCODING MEDIA STREAMS

2 H.323: Packet-based Multimedia Communications Systems

2.1 INTRODUCTION

2.2 H.323 STEP BY STEP

2.3 OPTIMIZING AND ENHANCING H.323

2.4 CONFERENCING WITH H.323

2.5 DIRECTORIES AND NUMBERING

2.6 H.323 SECURITY

2.7 SUPPLEMENTARY SERVICES

2.8 FUTURE WORK ON H.323

3 The Session Initiation Protocol (SIP)

3.1 THE ORIGIN AND PURPOSE OF SIP

3.2 OVERVIEW OF A SIMPLE SIP CALL

3.3 CALL HANDLING SERVICES WITH SIP

3.3.4 MULTIPARTY CONFERENCING

3.4 SIP SECURITY

3.5 INSTANT MESSAGING (IM) AND PRESENCE

4 The 3GPP IP Multimedia Subsystem (IMS) Architecture

4.1 INTRODUCTION

4.2 OVERVIEW OF THE IMS ARCHITECTURE

4.3 THE IMS CSCFS

4.4 THE FULL PICTURE: 3GPP RELEASE 8, TISPAN

5 The Media Gateway to Media Controller Protocol (MGCP)

5.1 INTRODUCTION: WHY MGCP?

5.2 MGCP 1.0

5.3 SAMPLE MGCP CALL FLOWS

5.4 THE FUTURE OF MGCP

6 Advanced Topics: Call Redirection

6.1 CALL REDIRECTION IN VOIP NETWORKS

7 Advanced Topics: NAT Traversal

7.1 INTRODUCTION TO NETWORK ADDRESS TRANSLATION

7.2 WORKAROUNDS FOR VOIP WHEN THE NETWORK CANNOT BE CONTROLLED

7.3 RECOMMENDED NETWORK DESIGN FOR SERVICE PROVIDERS

7.4 CONCLUSION

Annex

Index

This edition first published 2011 © 2011 John Wiley & Sons Ltd.

First edition published 2005

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Library of Congress Cataloguing-in-Publication Data

Hersent, Olivier. IP telephony: deploying VoIP protocols and IMS infrastructure/Olivier Hersent. — 2nd ed.

p. cm. Includes index.

ISBN 978-0-470-66584-8 (cloth)

1. Internet telephony. 2. Convergence (Telecommunication) I. Title. TK5105.8865.H47 20010 004.69'5-dc22

2010024553

A catalogue record for this book is available from the British Library.

Print ISBN 9780470665848 (H/B) ePDF ISBN: 9780470973264 oBook ISBN: 9780470973080

Abbreviations

3GPPThird Generation Partnership ProjectA-BGFAccess Border Gateway functionA-RACFAccess RACFA/VAudio-visualAADAverage Acknowledgement DelayAAL2ATM Adaptation Layer 2ACDAutomatic Call DistributionACELPAlgebraic-Code-Excited Linear-PredictionACFAdmission ConfirmACLAccess Control ListACMAddress Complete MessageADEVAverage Delay DeviationADPCMAdaptive Differential Pulse Mode ModulationADSLAsymmetric Digital Subscriber LineAESAdvanced Encryption StandardAFApplication FunctionAGCFAccess Gateway Control FunctionAMFAccess Management FunctionAMRAdaptive Multi-RateAMR-WBAdaptive Multi-Rate (Wide Band)AN-GWAccess Network GatewayANDSFAccess Network Discovery and Selection FunctionANMAnswer MessageANSIAmerican National Standard InstituteAOCAdvice of ChargeAoRSIP Address of RecordAPDUApplication Protocol Data UnitAPIApplication Programming InterfaceARFAccess Relay FunctionARJAdmission RejectARQAdmission RequestASApplication ServerASCIIAmerican Standard Code for Information InterchangeASFApplication Server FunctionASN-1Abstract Syntax Notation OneASPApplication Service ProviderASRAutomatic Speech Recognition or Answer Seizure RatioATMAsynchronous Transfer ModeAUCXAudit ConnectionAUEPAudit EndpointAVCAdvanced Video CodingAVTAudio/Video TransportB2BUABack-to-back User AgentBASICBeginners’ All-purpose Symbolic Instruction CodeBBERFBearer Binding and Event Reporting FunctionBCFBandwidth ConfirmBERBasic Encoding RuleBGCFBreakout Control Gateway FunctionBGFBorder Gateway FunctionBICCBearer Independent Call ControlBNFBackus-Naur FormBRJBandwidth RejectBRQBandwidth RequestBTFBasic Transport FunctionC-BGFCore Border Gateway functionC-RACFCore RACFCACall AgentCALEACommunication Assistance for Law Enforcement ActCallIDCall IdentifierCBCCipher Block ChainingCCCSRC CountCCFCharging Collector FunctionCCIRConsultative Committee for International Radio (ITU)CDMACode Division Multiplex AccessCDRCall Detail RecordCEDCallEDCELPCode-excited Linear PredictionCFBCipher FeedbackCFUCall Forwarding UnconditionalCICCircuit Identification CodeCIDConference IdentifierCIFCommon Intermediary FormatCLECCompetitive Local Exchange CarrierCLFConnectivity Session Location and Repository FunctionCLIPCalling Line Identity PresentationCLIRCalling Line Identity RestrictionCMACall Management AgentCMTSCable Modem Termination SystemCNDCustomer Network DeviceCNGCalliNG; Comfort Noise GeneratorCNGCustomer Network GatewayCNGCFCNG Configuration FunctionCOCentral OfficeCodecCOder DECoderCoIxConnectivity-oriented InterconnectionCOMEDIAConnection-oriented Media Transportin SDPCOPSCommon Open Policy ServiceCPECustomer Premises EquipmentCPGCall Progress (Message)CPIMCommon Profile for Instant MessagingCPLCall Processing LanguageCPNCustomer Premises NetworkCPUCentral Processing UnitCRCarriage ReturnCRCCyclic Redundancy CheckCRCXCreate ConnectionCRLFCarriage Return and Line FeedCRVCall Reference ValueCS-ACELPConjugate Structure, Algebraic Code-Excited Linear PredictionCSRCContributing SourceCTIComputer Telephony IntegrationDCFDisengage ConfirmDCMEDigital Circuit Multiplication EquipmentDCNDisconnectDCSDistributed Call SignalingDCTDiscrete Cosine TransformDDNSDynamic DNSDES/CBCData Encryption Standard, Cipher Block ChainingDESData Encryption StandardDHCPDynamic Host Configuration ProtocolDiffServDifferentiated ServicesDISDigital Identification SignalDLDownlinkDLCXDelete ConnectionDLSRDelay Since Last Sender ReportDNSDomain Name SystemDNSSECDomain Name System Security ProtocolDOCSISData over Cable Service Interface SpecificationDoSDenial of ServiceDRJDisengage RejectDRQDisengage RequestDSLDigital Subscriber LineDSMIPDual Stack Mobile IPDSPDigital Signal ProcessorDSS1Digital Subscriber Signaling 1DTMFDual-Tone Multi-FrequencyDTXDiscontinuous TransmissionDVMRPDistance Vector Multicast Routing ProtocolE-CSCFEmergency-CSCFE-UTRANEvolved-UTRANECBElectronic Code BookECFElementary Control FunctionEFFElementary Forwarding FunctionEFREnhanced Full RateENUM“Electronic Numbers” ProtocolEOLEnd of LineEOPEnd of ProcedureEPCFEndpoint Configuration CommandePDGevolved Packet Data GatewayETSIEuropean Telecommunications Standardisation InstituteETSI TIPHONETSI Telephony and Internet Protocol Harmonization Over NetworksETTBEthernet to the BuildingETTXEthernet to the <anything> (Curb, Home, Building)FCFFax Control FieldFCSFrame Check SequenceFECForward Error CorrectionFIFFax Information FieldFIFOFirst in First OutFIPS PUBFederal Information Processing Standards PublicationFRFull-rateFSFastStartGCFGatekeeper ConfirmGEFGeneric Extensibility FrameworkGGSNGateway GPRS Support NodeGKGatekeeperGOBsGroup of BlocksGRJGatekeeper RejectGRQGatekeeper RequestGSMGlobal System for Mobile CommunicationsGTDGlobal Transparency DescriptorGTPGeneric Tunneling ProtocolHDHang Down (off-hook)HDLCHigh-level Data Link ControlHLRHome Location RegisterHLR/AuCHLR Authentication CenterHRHalf RateHSPAHigh Speed Packet AccessHSSHome Subscriber ServerHTMLHypertext Markup LanguageHTTPHypertext Transfer ProtocolHUHang Up (on-hook)I-BGFInterdomain Border Gateway functionI-CSCFInterrogating Call/Session Control FunctionIADIntegrated Access DeviceIAMInitial Address MessageIANAInternet Assigned Numbers AuthorityIARIIMS Application Reference Identifier (IARI)IBCFInterconnection Border Control FunctionICIDIMS Charging IdentifierICMPInternet Control Message ProtocolICSIIMS Communication Service IdentifierIECISO International Electrotechnical CommissionIETFInternet Engineering Task ForceIFInterfaceIFPInternet Fax ProtocolIFTInternet Fax Transmission protocolILSInternet Locator Service (Microsoft)IMInstant MessagingIMCNIP Multimedia Core NetworkIMPIIP Multimedia Private IdentityIMPPInstant Messaging and Presence ProtocolIMPUIP Multimedia Public IdentityIMSIP Multimedia subsystemIMTCInternational Multimedia Teleconferencing ConsortiumINIntelligent NetworkINAPIntelligent Network Application ProtocolIntServIntegrated ServicesIOIInter Operator IdentifierIPIntelligent PeripheralIP CANInternet Protocol Connectivity Access NetworkIP-PBXInternet Protocol–Private Branch ExchangeIPDCInternet Protocol Device ControlIPRIntellectual Property RightsIPSecInternet Protocol SecurityIRCInternet Relay ChatIRQInformation RequestIRRInformation Request ResponseISDNIntegrated Service Digital NetworkISPInternet Service ProviderISUPISDN USER PART protocolITSPInternet Telephony Service ProviderITUInternational Telecommunications UnionIVRInteractive Voice ResponseIWFInterworking FunctionJFIFJPEG File Interchange FormatJPEGJoint Photographic Experts GroupLANLocal Area NetworkLCDLiquid Crystal DisplayLCFLocation ConfirmLD-CELPLow-delay, Code-excited Linear PredictionLDAPLightweight Directory Access ProtocolLFLine FeedLNPLocal Number PortabilityLRJLocation RejectLRQLocation RequestLSPLine Spectral PairLSRLast Sender ReportLTELong Term EvolutionMMarker Bit (RTP)mBoneMulticast Backbone of the InternetMCMultipoint ControllerMCFMessage ConfirmationMCUMultipoint Control UnitMD5Message Digest 5MDCXModify ConnectionMEGACOMedia Gateway ControllerMGCFMedia Gateway Control FunctionMGCPMedia Gateway Control ProtocolMGCP/LMGCP LineMGCP/TMGCP TrunkMGFMedia Gateway FunctionMHModified HuffmannMIMEMultipurpose Internet Mail ExtensionMIPMobile IPMIPSMillions of Instructions Per SecondMMEMobility Management EntityMMSMultimedia Message ServiceMMUSICMultiparty Multimedia Session ControlMOSMean Opinion ScoreMPMultipoint ProcessorMP-MLQMultipulse Maximum Likelihood QuantizationMPEGMoving Picture Experts GroupMPLSMultiprotocol Label SwitchingMRFCMedia (or Multimedia) Resource Function ControllerMRFPMedia (or Multimedia) Resource Function ProcessorMTPMessage Transfer PartMTTMinimum Transmission TimeMTUMaximum Transmission UnitMWIMessage Waiting IndicationMXMail ExchangeNACFNetwork Access Configuration FunctionNAPTNetwork Address and Port TranslationNAPTRNaming Authority Pointer RecordNASNetwork Access ServerNASSNetwork Attachment SubsystemNATNetwork Address TranslationNCSNetwork Based Call Signaling ProtocolNFENetwork Facility ExtensionNTFYNotifyNTPNetwork Time ProtocolNTSCNational Television System CommitteeOFBOutput FeedbackOGWOriginating GatewayOIDObject IdentifierOLCOpen Logical ChannelOOOn–offOSOperating SystemOSPOpen Settlement ProtocolP-CSCFProxy Call/Session Control FunctionP-framePrediction FramePALPhase-alternation-linePBDFProfile Data Base FunctionPBXPrivate Branch ExchangePCCPolicy and Charging ControlPCEFPolicy and Charging Enforcement FunctionPCMPulse Code ModulationPCMAPulse Code Modulation A LawPCMUPulse Code Modulation μ LawPCRFPolicy and Charging Rule FunctionPDFPolicy Decision FunctionPDN-GWPacket Data Network GatewayPDUProtocol Data UnitPEPPolicy Enforcement PointPERPacked Encoding RulesPESPSTN/ISDN Emulation SubsystemPGRPages Received (Fax)PGSPages Sent (Fax)PIProgress IndicatorPIDFPresence Information Data FormatPIMProtocol-independent MulticastPMIPProxy Mobile IPPOSIXPortable Open System InterconnectPOTSPlain Old Telephone ServicePSTNPublic Switched Telephone NetworkPTPayload TypeQCIFQuarter CIFV (144*176)QoPQuality of ProtectionQoSQuality of ServiceRACFResource and Admission Control FunctionRACSResource and Admission Control SubsystemRAIResource Availability IndicatorRANRadio Access NetworkRASRegistration, Admission, Status ProtocolRCReception Report CountRCEFResource Control Enforcement FunctionRCFRegistration ConfirmRDRestart DelayREDRandom Early DetectionRFCRequest for CommentsRGBRed–green–blueRGWResidential GatewayRLERun Length EncodingRMRestart MethodRQNTNotification RequestRRResource RecordRRJRegistration RejectRRQRegistration RequestRRsResource RecordsRSARivest, Shamir, Adleman (public keyalgorithm)RSIPRestart in ProgressRSTResetRSVPResource ReserVation ProtocolRTCReturn to CommandRTCPReal-time Control ProtocolRTORetransmission TimeoutRTP/AVTReal Time Protocol under the Audio/Video ProfileRTPReal-time ProtocolRTPReal-time Transport ProtocolRTSPReal-time Streaming ProtocolS-CSCFServing Call/Session Control FunctionS-GWServing GatewayS/MIME SecureMultipurpose Internet Mail ExtensionSAPSession Announcement ProtocolSBCSession Border ControllerSCNSwitched Circuit NetworkSCPService Control PointSCTPStream Control Transport ProtocolSDESSource Description RTP PacketSDLSpecification and Description LanguageSDPSession Description ProtocolSECAMSequentiel Couleur a MemoireSGCFSignaling Gateway Control FunctionSGCPSimple Gateway Control ProtocolSGFSignaling Gateway FunctionsimcapSimple Capability (SDP Declaration)SIMPLESIP for Instant Messaging and Presence Leveraging ExtensionsSIPSession Initiation ProtocolSIPSSession Initiation Protocol SecureSLFSubscription Locator FunctionSMGSpecial Mobile Group (of ETSI)SMSShort Message ServiceSMTPSimple Mail Transfer ProtocolSoIxService-oriented InterconnectionSPSingle SpaceSPDFService Policy Decision FunctionSQCIFSub-QCIF (128 × 96) Sender ReportSRSender ReportSRVServer DNS RecordSSSupplementary ServiceSS-CDSupplementary Service: Call DeflectionSS-CFBSupplementary Service: Call Forwarding on BusySS-CFNRSupplementary Service: Call Forwarding on No ReplySS-CFUSupplementary Service: Call Forwarding UnconditionalSS-DIVAll Diversion Supplementary ServicesSS7Signaling System 7SSFService Switching FunctionSSLSecure Sockets LayerSSWSoftswitchSTPSignaling Transfer PointSTUNSimple Traversal of UDP through Network Address TranslatorsSUDSingle Use DeviceTAPIMicrosoft Telephony APITCAPSS-7 Transaction CapabilitiesTCFTraining Check FunctionTCPTransport Control ProtocolTCSTerminal Capability SetTCS=0NullCapabilitySet Call Flow in H.323TDMTime Division MultiplexingTETerminal Equipment UnitTFTPTrivial File Transfer ProtocolTGCFTrunking Gateway Control FunctionTGWTerminating GatewayTIATelecommunications Industry Association (USA)TIPHONTelephony and Internet Protocol Harmonization over Networks (ETSI)TLSTransport Layer SecurityTLVType, Length, Value FormatTOTimeoutTPKTTransport Packet (RFC 1006)TTLTime to LiveTTSText to SpeechTURNTraversal Using Relay NATUASIP User AgentUAAFUser Access Authorization FunctionUCFUnregistration ConfirmUCSUniversal Character SetUDPUser Datagram ProtocolUDPTLUDP Transport LayerUEUser EquipmentUICCUniversal Integrated Circuit CardUIIUser Input IndicationULUplinkUMTSUniversal Mobile Telecommunication SystemUPSFUser Profile Server FunctionUPTUniversal Personal TelephonyURIUniform Resource IdentifierURJUnregistration RejectURLUniform Resource LocatorURNUniform Resource NameURQUnregistration RequestUSHUniversite de SherbrookeUTFUCS Transformation FormatUTRANUMTS Terrestrial Radio Access NetworkVADVoice Activity DetectorVASValue Added ServicesVASAValue Added Services AllianceVLANVirtual Local Area NetworkVoIPVoice over Internet ProtocolVPIMVoice Profile for Internet MessagingVPNVirtual Private NetworkVSELPVector Sum-excited Linear PredictionWAPWireless Application ProtocolWWWWorld Wide WebXMLeXtensible Markup LanguageXMPPeXtensible Messaging and Presence Protocol

Glossary

Abstract Syntax Notation-1 (ASN-1)Defined in ITU standard X.691.Access Control List (ACL)A packet filter on a router.Admission Confirm (ACF)A RAS message defined in H.225.0.Admission Reject (ARJ)A RAS message defined in H.225.0.Admission Request (ARQ)A RAS message defined in H.225.0.Application Protocol Data Units (APDUs)See H.450.1.Associate SessionA related session. Two related sessions must be synchronized (e.g., an audio session can specify a video session as being related). The receiving terminal must perform lip synchronization for those sessions.Backus–Naur Form (BNF)See RFC 2234.Bandwidth Confirm (BCF)A RAS message defined in H.225.0.Bandwidth Reject (BRJ)A RAS message defined in H.225.0.Bandwidth Request (BRQ)A RAS message defined in H.225.0.Basic Encoding Rule (BER)See ASN.1.Call Identifier (Call-ID)A globally unique call identifier.Call Reference Value (CRV)A 2-octet locally unique identifier copied in all Q.931 messages concerning a particular call (see also conference identifier).Conference Identifier (CID)This is not the same as the Q.931 Call Reference Value (CRV) or the call identifier (CID). The CID refers to a conference which is the actual communication existing between the participants. In the case of a multiparty conference, if a participant joins the conference, leaves, and enters again, the CRV will change, while the CID will remain the same.The Common Intermediary Format (CIF)A video format which has been chosen because it can be sampled relatively easily from both the 525-line and 625-line video formats: 352 × 288 pixelsContributing Source (CSRC)When an RTP stream is the result of a combination put together by an RTP mixer of several contributing streams, the list of the SSRCs of each contributing stream is added in the RTP header of the resulting stream as CSRCs. The resulting stream has its own SSRC.Disengage Confirm (DCF)A RAS message defined in H.225.0.Disengage Reject (DRJ)A RAS message defined in H.225.0.Disengage Request (DRQ)A RAS message defined in H.225.0.Dual-Tone Multi-Frequency (DTMF)Tones composed of two well-defined frequencies that represent digits 0–9, *, #. The combination of frequencies has been selected to be almost impossible to reproduce with the human voice. DTMF tones are used to dial from analog phones and to control IVR servers.Dynamic Host Configuration Protocol (DHCP)Used by end points to acquire a temporary IP address and important TCP/IP parameters (router IP address, DNS IP address, etc.) from a server in the network.End of Line (EOL)The end of line sequence for group 3 fax (001H).EnergyFor an image on a particular color, the sum of the squared color values of the pixels is called the energy.ENUMAn E.164 number resolution protocol defined in RFC 2916.Fast-ConnectA procedure to eliminate media delays after the connection of the call introduced in H.323v2. Another name used for the same procedure is Fast-Start.Fast-StartSee Fast-Connect.Gatekeeper Confirm (GCF)A RAS message defined in H.225.0.Gatekeeper Request (GRQ)A RAS message defined in H.225.0.Gatekeeper Reject (GRJ)A RAS message defined in H.225.0.Information Request (IRQ)A RAS message defined in H.225.0.Information Request Response (IRR)A RAS message defined in H.225.0.Initial Address Message (IAM)SS7 ISUP message initiating a call set-up.Inter-modeRefers to a video-coding mode where compression is achieved by reference to the previous, or sometimes the next, frame.Interactive Voice Response server (IVR)A machine accepting DTMF or voice commands, and executing some logic which interacts with the user using pre-recorded prompts or synthetic voice.Internet Fax Transmission (IFT)A protocol, see ITU recommendation T.38.Internet Relay Chat (IRC)The famous ’chat’ service of the Internet, based on a set of servers mirroring text-based conversations in real-time.Intra-modeRefers to a video-coding mode where compression is achieved locally (i.e., not relatively to the previous frame).IP-PBXPrivate phone switch with a VoIP wide area network interface. Most IP-PBXs have an H.323 WAN interface. See also IPBX.IPBXSame as IP-PBX. Some use the term IPBX for private phone switches which use only VoIP (i.e., the phones are also IP phones), whereas an IP-PBX can be a traditional PBX with analog phones and only uses a WAN VoIP interface. See IP-PBXJitterStatistical variance of packet interarrival time. It is the smoothed absolute value of the mean deviation in packet-spacing change between the sender and the receiver. The smoothing is usually done by averaging on a sliding window of 16 instantaneous measures.jitterVarying delay.Location Confirm (LCF)A RAS message defined in H.225.0.Location Reject (LRJ)A RAS message defined in H.225.0.Location Request (LRQ)A RAS message defined in H.225.0.macroblockFor the H.261 algorithm, a group of four 8*8 blocks.Maximum Transmission Unit (MTU)The largest datagram that can be sent over the network without segmentation.Multicast Backbone of the Internet (mBone)Capable of sending one packet to multiple recipients.Multipoint Control Unit (MCU)An H.323 callable end-point which consists of an MC and optional MPs.Multipoint Controller (MC)The H.323 which provides the control function for multiparty conferences.Multipoint Processor (MP)The H.323 entity which processes the media streams of the conference and does all the necessary switching, mixing, etc.Naming Authority Pointer Resource of the DNS (NAPTR)Defined in RFC 2915 and used notably by ENUM, see ENUM.Network Facility Extension (NFE)Defined in H.450.1.Network Time Protocol (NTP)This defines a standard way to format a timestamp, by writing the number of seconds since 1/1/1900 with 32 bits for the integer part and 32 bits for the decimal part expressed as number of 1/232 seconds (e.g., 0x800000000 is 0.5 s). A compact format also exists with only 16 bits for the integer part and 16 bits for the decimal part. The first 16 digits of the integer part can usually be derived from the current day, the fractional part is simply truncated to the most significant 16 digits.P-framePrediction frame obtained by motion estimation or otherwise, and representing only the difference between this image and the previous one.Packed Encoding Rules (PER)See ASN.1.Payload Type (PT)As defined by RTP.portAn abstraction that allows the various destinations of the packets to be distinguished on the same machine (e.g., Transport Selectors, or TSELs, in the OSI model, or IP ports). On the Internet, many applications have been assigned ’well-known ports’ (e.g., a machine receiving an IP packet on port 80 using TCP will route it to the web server).Prediction frame (P-frame)Obtained by motion estimation or otherwise, and representing only the difference between this image and the previous one.Private Branch Exchange (PBX)A private phone switch.Proxy serverAn intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it.Q-interface Signaling (QSIG)Protocol used at the Q-interface between two switches in a private network. ECMA/ISO have defined a set of QSIG standards.Real-time Control Protocol (RTCP)See RFC 1889.Real-time Transport Protocol (RTP)As specified by RFC 1889.-Registration Confirm (RCF)A RAS message defined in H.225.0.Registration Reject (RRJ)A RAS message defined in H.225.0.Registration Request (RRQ)A RAS message defined in H.225.0.Registration, Admission, and Status (RAS)The name of the protocol used between the gatekeeper and a terminal, and between gatekeepers for registration, admission, and status purposes. Defined in H.225.0.Return To Command (RTC)Six consecutive EOLs instructing a G3 Fax to return to command mode.Sender Report (SR)Used in RTCP and RTP.Session IDA unique RTP session identifier assigned by the master. The convention is that the value of the session ID is 1 for a primary audio session, 2 for a primary video session, and 3 for a primary data session. See Associate session.Single Use Device (SUD)See H.323 annex F.SIP dialogThis was defined in RFC 3261 as a peer-to-peer SIP relationship between two UAs which persists for some time. A dialog is established by SIP messages, such as a 2xx response to an INVITE request. A dialog is identified by a call identifier, a local tag, and a remote tag. A dialog was formerly known as a call leg in RFC 2543.SIP final responseA SIP response that terminates a SIP transaction (e.g., 2xx, 3xx, 4xx, 5xx, 6xx responses). See SIP provisional response.SIP provisional responseA SIP response that does not terminate a SIP transaction, as opposed to a SIP final response (1xx responses are provisional).SIP redirect serverA redirect server is a server that accepts a SIP request, maps the address into zero or more new addresses, and returns these addresses to the client. Unlike a proxy server, it does not initiate its own SIP request. Unlike a user agent server, it does not accept calls.SIP registrarA registrar is a server that accepts REGISTER requests. A registrar is typically co-located with a proxy or redirect server and may offer location services.SIP serverA server is an application program that accepts requests in order to service requests and sends back responses to those requests. Servers are either proxy, redirect, or user agent servers or registrars.SIP transactionA SIP transaction occurs between a client and a server, and comprises all messages from the first request sent from the client to the server up to a final (non-1xx) response sent from the server to the client. A transaction is identified by the CSeq sequence number within a single-call leg. The ACK request has the same CSeq number as the corresponding INVITE request, but comprises a transaction of its own.Stream Control Transport Protocol (SCTP)Defined in RFC 2960.Supplementary Services (SS-DIV)Includes all diversion supplementary services, such as SS-CFU, SS-CFB, SS-CFNR, SS-CD.Switched Circuit Network (SCN)A generic term for the ’classic’ phone network, including PSTN, ISDN, and GSM.Synchronization Source (SSRC)Source of an RTP stream, identified by 32 bits in the RTP header. All the RTP packets with a common SSRC have a common time and sequencing reference.TalkspurtA period during which a participant usually speaks, as opposed to silence periods.Time Division Multiplexing (TDM)The traditional voice transmission and switching technique based on assigning each communication a fixed “time slot” on a communication line between central offices.TPKTA TCP connection establishes a reliable data stream between two hosts, but there is no delimitation of individual messages within this stream. RFC 1006 defines a simple TPKT packet format to delimit such messages. It consists of a version octet (“3”), two reserved octets (“00”), and the total length of the message including the previous headers (2 octets).Transport addressCombination of a network address (e.g., IP address 10.0.1.2) and port (e.g., IP port 1720) which identifies a transport termination point.Transport Control Protocol (TCP)The most widely used, reliable transport protocol for IP networks.Transport Layer Security (TLS)A secure protocol using TCP, defined in RFC 2246.Trivial File Transfer Protocol (TFTP)A very simple file transfer protocol over UDP, frequently used by IP appliances to download their initial configuration parameters.Uniform Resource Identifier (URI)Defines a uniform syntax and semantic convention for any resource. The URI is defined in RFC 2396. See also URL, URN.Uniform Resource Locator (URL)A specific type of URI identifying a resource by its primary network address. URLs are used by SIP to indicate the originator, current destination, and final recipient of a SIP request, and to specify redirection addresses. See also URI.Uniform Resource Name (URN)A specific type of URI required to be universally unique and persistent even if the resource ceases to exist. See also URI.Unregistration Confirm (UCF)A RAS message defined in H.225.0.Unregistration Reject (URJ)A RAS message defined in H.225.0.Unregistration Request (URQ)A RAS message defined in H.225.0.User Agent Client (UAC)Also known as a calling user agent. A user agent client is a client application that initiates the SIP request.User Agent Server (UAS)Also known as a called user agent. A user agent server is a server application which contacts the user when a SIP request is received and returns a response on behalf of the user. The response accepts, rejects, or redirects the request.User agentA SIP end system participating in a SIP transaction. See UAC, UAS.User Datagram Protocol (UDP)The most widely used unreliable transport protocol for IP networks. UDP only guarantees data integrity by using a checksum, but an application using UDP has to take care of any data recovery task.-ZoneAn H.323 zone is the set of all H.323 end points, MCs, MCUs, and gateways managed by a single gatekeeper.

Preface

VoIP 1998–2004, 6 YEARS FROM R&D LABS TO LARGE SCALE DEPLOYMENTS

Since 1998 Voice over IP, in short VoIP, has been the favorite buzzword of the telecom industry. In 1998, IP was not yet as established and dominant as it is today, and most telecom engineers still believed that only ATM technology would be able to support multimedia applications. Indeed at this time most of us experienced the Internet only through dial-up modems and most ISPs, unable to keep-up with the exploding demand for Internet connections, were providing a level of service that could hardly qualify even for ‘best effort’.

But even in this context, the R&D teams that started to work on VoIP were not simply taking a leap of faith. Their bet on VoIP was backed by the last developments of packet networking theory, which proved that properly designed IP networks could provide an appropriate support for applications requiring quality of service. Knowing this, most of these teams felt confident that VoIP could be deployed on a wide scale in the future, and in the mean time tried to evaluate what could be the impact of VoIP, compared to previous technologies.