SIP based VoiceXML browser CINEMA (Columbia InterNet Extensible Multimedia Architecture) presented by – Kundan Singh, Joint work with Wenyu Jiang, Jonathan Lennox, Sankaran Narayanan, Henning Schulzrinne, Xiaotao Wu Columbia University, New York. More information at http://www.cs.columbia.edu/IRT/cinema/ Project Objectives A flexible architecture to support clients and servers for wide range of multimedia communication applications such as video conferencing, Internet telephony/radio, interactive voice response, unified messaging, presence and multimedia collaboration. Performance Sipstone: benchmark for SIP servers Different signaling vs. media components Black-box measurement and white-box profiling Load balancing, thread pooling, and reactive system to improve performance Novel peer-to-peer IP telephony using SIP Approach Develop protocols (SIP, RTSP, RTP,…) Implement common reusable libraries Provide distributed servers components Integrate with web, email, phone systems Session Initiation Protocol (SIP)-based enterprise VoIP infrastructure Load sharing and failover in SIP Internal Telephone Extn: 7040 SIP/PSTN Gateway Department PBX Web based configuration Web server switch SQL database sipd: Proxy, redirect, Registrar server NetMeeting H.323 rtspd: media server sipum: Unified messaging Quicktime RTSP clients RTSP 713x CINEMA servers sipconf: Conference server siph323: SIP-H.323 translator Local/long distance 1-212-5551212 PSTN SIP VXML vxml cgi example.com _sip._udp SRV 0 0 s1 SRV 0 0 s2 SRV 0 0 s3 SRV 1 0 ex M S sip:bob@example.com sip:bob@b.example.com s1 s2 s3 a1 a2 b1 b2 a.example.com SRV 0 0 a1 SRV 1 0 a2 b.example.com SRV 0 0 b1 SRV 1 0 b2 P P2P VoIP using SIP Unified messaging using SIP and RTSP Peer-to-peer Internet telephony avoids the configuration and maintenance cost of server-based infrastructure and dependency on controlled infrastructure such as DNS. We use Chord algorithm on top of SIP for an interoperable, scalable and robust P2P-SIP endpoint. Master Web scripts D2 P2 D1 P1 phone.cs.columbia.edu sip2.cs.columbia.edu REGISTER proxy1 = phone.cs backup = sip2.cs _sip._udp SRV 0 0 5060 phone.cs.columbia.edu SRV 1 0 5060 sip2.cs.columbia.edu replication Presence and event notification PA registrar Presence server office.com SUBSCRIBE NOTIFY REGISTER alice@home.com bob@office.com PUA PUA + PA Multimedia conferencing sipc sipconf SIP323 Netmeeting SIP/PSTN SIP H.323 Netmeeting e*phone sipd sip323 SIP-H.323 gateway A SIP/RTP-based centralized conference server to support audio mixing, video forwarding, text chat and screen sharing among heterogeneous endpoints such as PC and phones. It has play-out delay adjustment for wide area Internet, web-based conference setup, high quality audio (G.722, G.711) as well as low bit rate codecs (GSM, DVI). A signaling translator between ITU-T’s multistage H.323 and IETF’s SIP that supports different dialing modes, has a built-in gatekeeper and is transparent to media path. Overview Multimedia communication Audio, video, text, screen sharing, … PSTN interworking, IVR Multi-devices IP-phone, telephone, X10, Ncast, … Collaboration Voicemail, discussion forum,… Multimedia application components Program Call routing SIP SAP RSVP RTCP RTP Media G.711 MPEG RTSP Signaling Quality of service Media transport Internet Telephony Radio/TV Messaging and Presence Interactive voice response Unified messaging Video conferencing Physical layer Link layer Network (IPv4, IPv6) Transport (TCP, UDP) Application layer Voice XML DTMF Mixing Speech/ text SDP SIP “forking” proxy Programmable IP telephony services PSTN SIP user agent SIP/PSTN gateway Web server CGI, servlet, JSP SIP based VoiceXML browser SIP phone              Media server Call Request Fetch VoiceXML pages Get streaming media Press 1 to listen to next message, 2 to forward … Interactive voice response Programmable call routing based on time of day, caller id, etc., using server side Call processing language, Common Gateway interface (CPL), Java servlets or client side Language for End System services (LESS) scripts Transport layer (TCP/UDP) RTP Interface HTTP Message Parsing RTSP transaction SIP transaction Client Branch RTSP API RTSP server SIPUA API SIP proxy Other Applications Libraries (C/C++) SIP, RTP, audio mixing, DB interface, SNMP interface, RTSP, DNS SRV, win32 portability,… PSTN interworking sip:wenyu@cs.columbia.edu Telephone network SIP/PSTN gateway SIP server IP endpoint subscriber +1 212 9397040 sip:7141@cs.columbia.edu … moving from IP telephony to real-time multimedia collaboration… Layered Architecture