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 telephony Internet radio/TV 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

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

Personal mobility 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

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

Programmability Common gateway interface (CGI) 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)

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

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

Enterprise VoIP PSTN CINEMA servers Telephone switch Local/long distance e.g., 1-212-5551212 sipconf: conference server rtspd: media 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 server Web based configuration SIP/PSTN Gateway e.g., Cisco 2600 vxml SIP VXML 7134 7136 siph323: SIP-H.323 translator H.323 alice@cs.columbia.edu (software phone) H.323 clients e.g., NetMeeting

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

VoIP and wireless Improvement in our solution

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

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

Serverless (P2P) VoIP Server-based Peer-to-peer P2P-SIP 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 C S P

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

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

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

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

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