SIP-based VoiceXML browser (sipvxml) CINEMA (Columbia InterNet Extensible Multimedia Architecture) presented by – Kundan Singh, Joint work with Wenyu Jiang, Jonathan Lennox, Sankaran Narayanan, Henning Schulzrinne, Xiaotao Wu 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 example.com _sip._udp SRV 0 0 s1 SRV 0 0 s2 SRV 0 0 s3 SRV 1 0 ex Second stage proxy/registrar (sipd) CINEMA servers P P2P VoIP using SIP First stage stateless proxy server farm Unified messaging using SIP and RTSP M rtspd: media server Local/long distance e.g., 1-212-5551212 sipconf: conference server a1 a.example.com _sip._udp SRV 0 0 a1 SRV 1 0 a2 Telephone switch 1 2 5 4 7 6 3 RTSP S RTSP clients e.g., Quicktime s1 a2 PSTN Department PBX sipd Bob’s phone Peer-to-peer Internet telephony avoids the configuration and maintenance cost of server-based architecture 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. sipum: unified messaging sip:bob@example.com Internal Telephone e.g., 7040 sipd: proxy, redirect, registrar sipum s2 sip:bob@b.example.com Alice’s phone 713x M SQL database cgi Web server s3 b1 b.example.com _sip._udp SRV 0 0 b1 SRV 1 0 b2 Alice (caller) calls Bob The SIP server forks the call to Bob’s phone and the mail server After 10 seconds, the mail server sets up RTSP sessions to playback welcome message and to record mail Mail server accepts the call SIP server cancels the other branch SIP server forwards the acceptance Media packets are sent directly between the RTSP server and caller vxml S SIP/PSTN Gateway e.g., Cisco 2600 Web based configuration b2 rtspd SIP VXML Presence and event notification Web scripts Web scripts 7134 7136 H.323 office.com D1 D2 bob@office.com siph323: SIP-H.323 translator Master Slave Presence server Bi-directional replication Slave Master alice@cs.columbia.edu (software phone) PA PUA H.323 clients e.g., NetMeeting alice@home.com REGISTER P1 P2 SUBSCRIBE Multimedia conferencing NOTIFY PUA registrar gatekeeper SIP-H.323 gateway phone.cs.columbia.edu sip2.cs.columbia.edu sipd 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). PUA + PA REGISTER Low bitrate SIP323 SIP H.323 _sip._udp SRV 0 0 5060 phone.cs.columbia.edu SRV 1 0 5060 sip2.cs.columbia.edu High quality proxy1 = phone.cs backup = sip2.cs 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 sipconf Interactive voice response SIP/PSTN Internet Telephony Internet Radio/TV Programmable SIP proxy Messaging and Presence Programmable IP telephony services Unified messaging Interactive voice response (IVR) Video conferencing PSTN phone 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 SIP/PSTN gateway vxml cgi CPL SIP SAP RSVP RTCP Media G.711 MPEG SQL Fetch VoiceXML pages Call request Web server RTSP CGI, servlet, JSP RTP Other Applications Application layer SIP phone              Get streaming media Transport (TCP, UDP) SIP-based VoiceXML browser (sipvxml) Libraries (C/C++) SIP, RTP, audio mixing, DB interface, SNMP interface, RTSP, DNS SRV/NAPTR, win32 portability,… Transport layer (TCP/UDP) RTP Interface HTTP Message Parsing RTSP transaction SIP transaction Client Branch RTSP API RTSP server SIPUA API SIP proxy Quality of service Media transport Network (IPv4, IPv6) Signaling Press 1 to listen to next message, 2 to forward … Media server Link layer SIP phone Physical layer Program Call routing Voice XML DTMF Mixing Speech/ text SDP PSTN interworking IP endpoint +1 212 9397040 … moving from IP telephony to real-time multimedia collaboration… sip:wenyu@cs Telephone network Telephone subscriber SIP/PSTN gateway SIP server (sipd) Layered Architecture sip:7141@cs.columbia.edu