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Internet Telephony

Internet Telephony. Shivkumar Kalyanaraman Based upon slides of Henning Schulzrinne (Columbia). Overview. Telephony: history and evolution IP Telephony: Why ? Adding interactive multimedia to the web Being able to do basic telephony on IP with a variety of devices

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Internet Telephony

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  1. Internet Telephony Shivkumar Kalyanaraman Based upon slides of Henning Schulzrinne (Columbia)

  2. Overview • Telephony: history and evolution • IP Telephony: Why ? • Adding interactive multimedia to the web • Being able to do basic telephony on IP with a variety of devices • Basic IP telephony model • Protocols: SIP, H.323, RTP, Coding schemes, MGCP, RTSP • Future: Invisible IP telephony and control of appliances

  3. Public Telephony (PSTN) History • 1876 invention of telephone • 1915 first transcontinental telephone (NY–SF) • 1920’s first automatic switches • 1956 TAT-1 transatlantic cable (35 lines) • 1962 digital transmission (T1) • 1965 1ESS analog switch • 1974 Internet packet voice • 1977 4ESS digital switch • 1980s Signaling System #7 (out-of-band) • 1990s Advanced Intelligent Network (AIN)

  4. Telephone Service in the US AT&T Divestiture

  5. Telephone System Overview • Analog narrowband circuits: home-> central office • 64 kb/s continuous transmission, with compression across oceans • -law: 12-bit linear range -> 8-bit bytes • Everything clocked a multiple of 125 s • Clock synchronization framing errors • AT&T: 136 “toll”switches in U.S. • Interconnected by T1, T3 lines & SONET rings • Call establishment “out-of-band” using packet-switched signalingsystem (SS7)

  6. Telephony: Multiplexing • Telephone Trunks between central offices carry hundreds of conversations: Can’t run thick bundles! • Send many calls on the same wire: multiplexing • Analog multiplexing • bandlimit call to 3.4 KHz and frequency shift onto higher bandwidth trunk • Digital multiplexing: convert voice to samples • 8000 samples/sec => call = 64 Kbps

  7. Trends: Price of Phone Calls: NY - London AT&T Divestiture

  8. Trends: Data vs Voice Traffic Since we are building future networks for data, can we slowly junk the voice infrastructure and move over to IP?

  9. Trends: Phone vs Data Revenues

  10. Private Branch Exchange (PBX) Post-divestiture phenomenon... 7040 212-8538080 External line 7041 Telephone switch Corporate/Campus Private Branch Exchange Another switch 7042 7043 Corporate/Campus LAN Internet

  11. IP Telephony: PBX Replacement Another campus Corporate/Campus 7040 8151 External line 8152 7041 PBX PBX 8153 7042 8154 7043 Internet LAN LAN

  12. Voice over Packet Market Forecast – North America

  13. Invisible Internet Telephony • VoIP technology will appear in . . . • Internet appliances • home security cameras, web cams • 3G mobile terminals • fire alarms • chat/IM tools • interactive multiplayer games

  14. IPtel for appliances: “Presence”

  15. Speech Coders Waveform Coders Source Coders Time Domain: PCM, ADPCM Frequency Domain: e.g. Sub-band coder, Adaptive transform coder Linear Predictive Coder Vocoder Taxonomy of Speech Coders • Waveform coders:attempts to preserve the signal waveform not speech specific. • PCM 64 kbps, ADPCM 32 kpbs, CVSDM 32 kbps • Vocoders: • Analyse speech, extract and transmit model parameters • Use model parameters to synthesize speech • LPC-10: 2.4 kbps • Hybrids: Combine best of both… Eg: CELP

  16. Our focus VoIP Camps Circuit switch engineers “We over IP” “Convergence” ITU standards Conferencing Industry Netheads “IP over Everything” H.323 SIP “Softswitch” BICC ISDN LAN conferencing I-multimedia WWW Call Agent SIP & H.323 BISDN, AIN H.xxx, SIP IP IP IP “any packet”

  17. Internet Multimedia Protocol Stack

  18. IP Telephony Protocols: SIP, RTP • Session Initiation Protocol - SIP • Contact “office.com” asking for “bob” • Locate Bob’s current phone and ring • Bob picks up the ringing phone • Real time Transport Protocol - RTP • Send and receive audio packets

  19. Internet Telephony Protocols: H.323

  20. H.323 (contd) • Terminals, Gateways, Gatekeepers, and Multipoint Control Units (MCUs)

  21. H.323 vs SIP Typical UserAgent Protocol stack for Internet Terminal Control/Devices Terminal Control/Devices H.245 RTCP Q.931 RAS RTCP SIP SDP Codecs Codecs RTP RTP TPKT TCP UDP Transport Layer IP and lower layers

  22. SIP vs H.323 • Binary ASN.1 PER encoding • Sub-protocols: H.245, H.225 (Q.931, RAS, RTP/RTCP), H.450.x... • H.323 Gatekeeper • Text based request response • SDP (media types and media transport address) • Server roles: registrar, proxy, redirect - Both use RTP/RTCP over UDP/IP - H.323 perceived as “heavyweight”

  23. Light-weight signaling: Session InitiationProtocol (SIP) • IETF MMUSIC working group • Light-weight generic signaling protocol • Part of IETF conference control architecture: • SAP for “Internet TV Guide” announcements • RTSP for media-on-demand • SDP for describing media • others: malloc, multicast, conference bus, . . . • Post-dial delay: 1.5 round-trip time (with UDP) • Network-protocol independent: UDP or TCP (or AAL5 or X.25)

  24. SDP: Session Description Protocol • Not really a protocol– describes data carried by other protocols • Used by SAP, SIP, RTSP, H.332, PINT. Eg: v=0 o=g.bell 877283459 877283519 IN IP4 132.151.1.19 s=Come here, Watson! u=http://www.ietf.org e=g.bell@bell-telephone.com c=IN IP4 132.151.1.19 b=CT:64 t=3086272736 0 k=clear:manhole cover m=audio 3456 RTP/AVP 96 a=rtpmap:96 VDVI/8000/1 m=video 3458 RTP/AVP 31 m=application 32416 udp wb

  25. SIP functionality • IETF-standardized peer-to-peersignaling protocol (RFC 2543): • Locate user given email-style address • Setup session (call) • (Re)-negotiate call parameters • Manual and automatic forwarding • Personal mobility:different terminal, same identifier • Call center: reach first (load distribution) or reach all (department conference) • Terminate and transfer calls

  26. SIP Addresses Food Chain

  27. SIP components • UAC:user-agent client (caller application) • UAS:user-agent server à accept, redirect, refuse call • redirect server:redirect requests • proxy server:server + client • registrar:track user locations • user agent = UAC + UAS • often combine registrar + (proxy or redirect server)

  28. IP SIP Phones and Adaptors 1 • Are true Internet hosts • Choice of application • Choice of server • IP appliances • Implementations • 3Com (3) • Columbia University • MIC WorldCom (1) • Mediatrix (1) • Nortel (4) • Siemens (5) Analog phone adaptor 2 3 Palm control 4 4 5

  29. rtspd Quicktime RTSP media server RTSP sipconf Telephone RTSP clients SIP conference server sipum Telephone switch SIP/RTSP Unified messaging Web based configuration sipd T1/E1 RTP/SIP SIP proxy, redirect server SQL database e*phone Cisco 2600 gateway Hardware Internet (SIP) phones sipc Web server SIPH.323 convertor NetMeeting sip323 Software SIP user agents H.323 SIP-based Architecture

  30. Web based configuration sipd Call Bob SIP proxy, redirect server SQL database e*phone Hardware Internet (SIP) phones sipc Web server Software SIP user agents Example Call • Bob signs up for the service from the web as “bob@ecse.rpi.edu” • sipd canonicalizes the destination to sip:bob@ecse.rpi.edu • He registers from multiple phones • sipd rings both e*phone and sipc • Bob accepts the call from sipc and starts talking • Alice tries to reach Bob • INVITE ip:Bob.Wilson@ecse.rpi.edu ecse.rpi.edu

  31. PBX Gateway PSTN Internal T1/CAS (Ext:7130-7139) External T1/CAS Call 9397134 Call 7134 Ethernet 1 2 4 5 3 Regular phone (internal) SIP server SQL database sipd sipc Bob’s phone 7134 => bob PSTN to IP Call

  32. PBX Gateway (10.0.2.3) PSTN External T1/CAS Internal T1/CAS Call 5551212 Call 85551212 Ethernet 4 5 2 3 1 5551212 Bob calls 5551212 Regular phone (internal, 7054) SIP server SQL database sipd sipc Use sip:85551212@10.0.2.3 IP to PSTN Call

  33. Dial 853-8119 Phone is ringing Disconnect Alice 939-7063 Bob 853-8119 .. The person is not available now please leave a message ... ... Your voice message ... Traditional voice mail system Bob can listen to his voice mails by dialing some number.

  34. Bob INVITE bob@phone1.office.com phone1.office.com INVITE bob@office.com REGISTER bob@vm.office.com Alice sipd INVITE bob@vm.office.com SIP-based Voicemail Architecture vm.office.com The voice mail server registers with the SIP proxy, sipd Alice calls bob@office.com through SIP proxy. SIP proxy forks the request to Bob’s phone as well as to a voicemail server.

  35. Bob phone1.office.com; CANCEL 200 OK Alice sipd 200 OK RTP/RTCP SETUP Voicemail Architecture v-mail vm.office.com; After 10 seconds vm contacts the RTSP server for recording. vm accepts the call. Sipd cancels the other branch and ... rtspd ...accepts the call from Alice. Now user message gets recorded

  36. SIP-H.323: Interworking ProblemsEg: Call setup translation H.323 SIP Q.931 SETUP INVITE Destination address (Bob@office.com) Q.931 CONNECT 200 OK Terminal Capabilities Media capabilities (audio/video) Terminal Capabilities ACK Open Logical Channel Media transport address (RTP/RTCP receive) Open Logical Channel • H.323: Multi-stage dialing

  37. MGCP and Megaco • Media Gateway Controller Protocol (RFC 2705) • Controlling Telephony Gateways from external call control elements called media gateway controllers (MGC) or call agents • Gateways: Eg: RGW : physical interfaces between VoIP network and residences • Call control "intelligence" is outside the gateways and handled by external call control elements • Goal: scalable gateways between IP telephony and PSTN • Successor to MGCP: H.248/Megaco

  38. MGCP Architecture Goal: large-scale phone-to-phone VoIP deployments RGW: Residential Gateway TGW: Trunk Gateway

  39. Summary • Telephony and IP Telephony • Protocols: SIP, SDP, H.323, MCGP • Example operation and services: • Calls, voice mail etc • Future: Integration with Web and long-term replacement for current telephone systems

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