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UT Dallas CS 6390 - 18. VoIP

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Slide 1VoIP vs IP TelephonyPublic Switched Telephony Network (PSTN)PSTN—StructureComparisonPacketizing VoiceVoIP ComponentsVoIP Use CasesVoIP StandardsVoIP Signaling ProtocolsSlide 11H.323H.323 Protocol StackH.323 ArchitectureSample H.323 Deployment ArchitectureH.323 Architecture ElementsH.323 Architecture ElementsH.323 Architecture ElementsH.323 Architecture ElementsH.323 AddressesSlide 21SIPHow to Address SIP EntitiesSIP Protocol MethodsSIP Architecture ElementsSIP Architecture ElementsSIP Network ArchitectureSIP Registration ExampleSIP Registration Example – 2SIP Call Setup & Tear DownSlide 31IAX2Slide 33GatewaysPSTN to VoIP address conversion?Using ENUM protocol: PSTN to SIPSlide 37PC Applications – PC to PCPSTN Bypass – PC to PhonePSTN Toll Bypass – Phone to PhonePSTN Bypass – Enterprise IP PBXPSTN Bypass – Broadband VoIP Service ProviderCable IP Telephony – Converged IP ArchitectureVoIP TransportCS 6390 – Advanced Computer Networks VoIP & IP TelephonySources: Several slides adopted from presentations by Niccolini, McCrum and ZhaoVoIP vs IP Telephony•Both deal with multimedia communication over IP networks–VoIP: digitize and carry analog speech data on IP network–IP Telephony: telephony service over IP with all related services (phone/fax/call forwarding/etc)•An alternative to PSTN•Will ignore the difference in our discussionPublic Switched Telephony Network (PSTN)•Traditional circuit switched based telephony network•Uses SS7 as its signaling protocol–Carried out of band on dedicated channel in PSTN•Fast call setup •Avoid fraudulent use – closed to access by users–Functions•set up, manage, and tear down calls•Caller ID, call forwarding as additional servicesPSTN—Structure Trunk Network Trunk NetworkCircuitSwitchCircuitSwitchPBXSignalingNetwork(SS7)SignalingNetwork(SS7)Voice Channel MultiplexingResidentialcustomersBusinesscustomersAccess linesComparisonNetwork Features PSTN (Voice) Internet (Data)Switch Circuit Switched Packet SwitchedConnection Connection Oriented ConnectionlessBit Rate Fixed and low <=64kb/s Wide variation to Gb/sBursts Nonexistent High (100/1000:1)Error tolerance Error free User error toleranceInfo resending Can not (real time) Can be done very fastDelay Must be low and stable Can be high and varyPacketizing VoiceVoIP Issues SolutionsEncoding delay Sophisticated codecsPacketization delay Small packets for VoIPNetwork delay Priorities and jitter buffers“High” bit-rate voice (64kb/s) Voice compressionConstant-bit-rate voice (PMC) Silence suppression and comfort noiseResend due to errors Real-time Transport Protocol (RTP)VoIP Components•Signaling – to establish & manage a call–SIP, H.323, MGCP & MEGACO/H248, IAX2•Encoding/decoding (codec) –voice is analog… need to digitize–G.7xx, GSM, H.26x, iLBC, Speex•Transport – how to carry digitized voice data–Packetize using RTP & RTCP protocols•Translation – VoIP to PSTN interoperability–Done by media/signaling gateways–Will see VoIP deployment scenariosVoIP Use CasesVoIP StandardsVoice Coding Methods (ITU-T G series, etc.)Call Control/Session Initialization (H.225, H.245, SIP)Timing (RTP)Gateway/GatekeeperControl (GLP, MGCP)Reliable/UnreliableTransport services (TCP/UDP)Internet Protocol (IP)Packet Network InfrastructureApplication LayerTransport LayerNetwork interface LayerNetwork LayerVoIP Signaling Protocols•H.323, SIP, MGCP & MEGACO/H248, IAX2•The role of signaling protocols–User location: caller has to find the location of the callee–Session establishment: callee decides on accept/reject/forward the call–Session negotiation: parties agree on parameters of the call–Call participation mgmt: handle events during the callH.323H.323•Packet-based multimedia communications systems – an ITU-T standard•Three types of interactions–Administrative operations•Registration, Admission, and Status (RAS) protocol–Signaling protocol: H.225–Capabilities negotiation & call control: H.245•Services–Local: call history, address book, etc–Nw-based: authorization, call admission–Supplementary: features to enhance user experience•Call forwardingH.323 Protocol StackH.323 ArchitectureSample H.323 Deployment ArchitectureH.323 Architecture Elements•Terminals – end points–HW clients–SW clients–Support protocols•H.323 protocols for signaling•Audio (and video) codecs•RTP for transport–Assigned to one or more aliases (username/URI) and/or phone number(s)H.323 Architecture Elements•Gateways–Interfaces b/w VoIP and PSTN (or other voice nws)–Translates the formats•Voice/data format – Media Gateway•Signaling format – Signaling GatewayH.323 Architecture Elements•Gatekeepers–Manage zones (H.323 entities in a domain)•End point registration•Address translation (phone number E.164 ID to TCP/IP addresses – IPaddr+port number)•Call routing•Admission control•Bandwidth management•Authorization control•etcH.323 Architecture Elements•Multipoint Control Unit (MCU)–Multipoint conferencing serverH.323 Addresses•Alias addresses–Easier to remember w.r.t. a network address–Corresponds to a unique nw addr: IP + port no–Must be unique within a zone•Eg: ksarac•Not necessarily unique globally•Need extra configuration for inter-zone call support–Eg: sip:[email protected] – globally uniqueSIPSIP•A protocol for creating, modifying, and terminating multimedia sessions w/one or more participants–Independent of underlying transport protocol•Supports name mapping and redirection services transparently•Basic scope of SIP is to exchange–IP addresses–Port numbersHow to Address SIP Entities•E-mail style addressing•Each user has globally reachable address–Bind to this address using REGISTER method–Used to establish a session•Example:–sip:[email protected] Protocol Methods•REGISTER – to bind permanent address to current location•INVITE – to invite a user to a call•ACK – to acknowledge messages•BYE – to terminate a connection•CANCEL – cancels a pending INVITE•OPTIONS – capability inquirySIP Architecture Elements•User agents (UA) – make/receive calls–UA Client (UAC) – initiates a SIP request–UA Server (UAS) – receives & responds to a request–A SIP end point can act as both UAC and UASSIP Architecture Elements•Network servers–Proxy servers •Receive UA request & decide which server to forward•Request may traverse


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UT Dallas CS 6390 - 18. VoIP

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