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IP telephony overview and demonstration

IP telephony overview and demonstration. Prof. Henning Schulzrinne (presented by Andrea Forte, Ron Shacham, Sangho Shin, Kundan Singh and Xiaotao Wu) http://www.cs.columbia.edu/IRT. Research topics in IRT lab. Internet radio/TV. Internet telephony. Peer-to-peer systems.

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IP telephony overview and demonstration

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  1. IP telephony overview and demonstration Prof. Henning Schulzrinne (presented by Andrea Forte, Ron Shacham, Sangho Shin, Kundan Singh and Xiaotao Wu) http://www.cs.columbia.edu/IRT

  2. Research topics in IRT lab Internet radio/TV Internet telephony Peer-to-peer systems Quality of service Security Internet Real Time Internet service discovery Content distribution VoIP and wireless Resource reservation Wireless ad hoc networks

  3. What is IP telephony? • Phone call + Internet • User identifier • Session Initiation Protocol – SIP office.com alice@columbia.edu Bob (2) (1) pc4.columbia.edu (3) home.com columbia.edu

  4. Personal mobility yahoo.com tel:12129397000 alice_95@yahoo.com Home alice@cs.columbia.edu tel:17185551234 Mobile desk@cs.columbia.edu cs.columbia.edu Alice.Cueba@cs.columbia.edu host.cs.columbia.edu Office

  5. Call setup Media path Control path Office Forking Mobile Visiting university Redirect Bob columbia.edu Home

  6. Programmability Double ringing sound when boss calls… Enter your authentication PIN for billing… Use finger for locating user… B2BUA Endpoint Make call when boss is online … Proxy/registrar Endpoint Forward to office phone during day, and home phone during evening… • Common gateway interface (CGI) • Call processing language (CPL) • SIP servlet • Language for End System Services (LESS)

  7. Server Proxy, register, redirect. Conferencing. Voicemail, IVR. IP phones Clients and servers Hardware phones Urgent SIP server Phone script Low-priority Voicemail Software phones

  8. Interworking with PSTN x7040 sip:bob@cs (212)5551212 • Translating: • Audio – better codecs on IP • Signaling – some features are lost • Identifiers – phone numbers • Determining transition points Telephone network (PSTN) PBX Telephone subscriber SIP/PSTN gateway SIP server IP endpoint

  9. SIP VXML Web server Enterprise VoIP CINEMA servers Telephone switch rtspd: media server Local/long distance e.g., 1-212-5551212 sipconf: conference server PSTN RTSP RTSP clients e.g., Quicktime Department PBX sipum: unified messaging Internal Telephone e.g., 7040 sipd: proxy, redirect, registrar 713x SQL database cgi Web based configuration vxml SIP/PSTN Gateway e.g., Cisco 2600 7134 7136 siph323: SIP-H.323 translator H.323 alice@cs.columbia.edu (software phone) H.323 clients e.g., NetMeeting

  10. Which wireless network? 802.11a/b/g Infrastructure mode (security) Ad-hoc mode What is handoff? Handoff happens when a mobile node moves beyond the radio range of one access point and enters another. Internet VoIP and wireless

  11. What is the problem? L2 Handoff time is too big (~500 ms) for seamless VoIP sessions (90 ms). VoIP and wireless

  12. Improvement in our solution VoIP and wireless

  13. Session Mobility • Focus on communication media: audio, video, instant messaging • Location sensors and presence, along with service discovery yields a list of local devices • Seamlessly transfer an active session between devices • Transfer all media to a single device or split over multiple devices • Privacy: keep audio on handset, watch video on large screen • Take advantage of benefits of different devices

  14. Local Devices Transcoder Internet SLP DA SLP UA SLP SA SIP SM SIP UA SIP UA Correspondent Node (CN) SLP SIP RTP SIP SM SIP UA SLP UA Mobile Node (MN) Session Mobility

  15. C C P P S C C P P C P Serverless (P2P) VoIP • Server-based • Cost: maintenance, configuration • Central points of failures • Controlled infrastructure (e.g., DNS) • Peer-to-peer • Robust: no central dependency • Self organizing, no configuration • Scalability • P2P-SIP • Efficient, interoperable, hybrid • Prototype implementation

  16. Summary • SIP-based architecture • Heterogeneous endpoints • Telephone, SIP phone, H.323 • Devices like lamp, video encoder • Multimedia collaboration • Conference, IM, discussion board, voicemail, file sharing • Advanced services • Programmable call routing, voice mail, interactive voice response • Fast handoff for WirelessLAN • P2P-SIP for serverless VoIP

  17. Conferencing sipc sipconf e*phone SIP/PSTN • Web configuration • Audio mixing • Video replication • SIP, PSTN or H.323

  18. Voicemail and IVR • Multi-platform (phone, PC) access • Standard based (SIP, RTSP) • Programmable dialogues

  19. Bob is in conf Turn on light You are In conf What’savailable Turn on conf’s light sip:conf Location NOTIFY Room conf Location agent Device GW SLinke Proxy LS Bob Trigger an action X10 sip:conf_pingtel for audio iButton reader SLP DA RFID reader SLP SA Resource discovery Tracking Location-basedServices in our lab

  20. INVITE sip:anyone_roomconf Room conf Location-basedServices in our lab Location agent Device GW SLinke Bob is in conf Turn on light Proxy LS Bob X10 You are In conf sip:conf_pingtel for audio What’savailable Turn on conf’s light iButton reader SLP DA RFID reader sip:conf SLP SA Guard communication behavior ‘Talk’ to alocation Location NOTIFY

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