1. 1.Alice (caller) calls Bob 2.The SIP server forks the call to Bob’s phone and the mail server 3.After 10 seconds, the mail server sets up RTSP sessions to playback welcome message and to record mail 4.Mail server accepts the call 5.SIP server cancels the other branch 6.SIP server forwards the acceptance 7.Media packets are sent directly between the RTSP server and caller 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/ example.com _sip._udp SRV 0 0 s1 SRV 0 0 s2 SRV 0 0 s3 SRV 1 0 ex M S M S sip:bob@example.com sip:bob@b.example.com s1 s2 s3 a1 a2 b1 b2 a.example.com _sip._udp SRV 0 0 a1 SRV 1 0 a2 b.example.com _sip._udp SRV 0 0 b1 SRV 1 0 b2 Unified messaging using SIP and RTSP 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. Internal Telephone e.g., 7040 SIP/PSTN Gateway e.g., Cisco 2600 Department PBX Web based configuration Web server Telephone switch SQL database sipd: proxy, redirect, registrar H.323 rtspd: media server sipum: unified messaging RTSP clients e.g., Quicktime RTSP 713x CINEMA servers sipconf: conference server siph323: SIP-H.323 translator Local/long distance e.g., 1-212-5551212 PSTN SIP VXML vxml cgi 7134 7136 alice@cs.columbia.edu (software phone) Approach  Develop protocols (SIP, RTSP, RTP,…)  Implement common reusable libraries  Provide distributed servers components  Integrate with web, email, phone systems 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 … moving from IP telephony to real-time multimedia collaboration… 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. Multimedia conferencing 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). Layered Architecture Other Applications Session Initiation Protocol (SIP)-based enterprise VoIP infrastructure SIP phone SIP/PSTN gateway Web server CGI, servlet, JSP SIP-based VoiceXML browser (sipvxml) 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 (IVR) Overview  Multimedia communication Audio, video, text, screen sharing, … PSTN interworking, IVR  Multi-devices IP-phone, telephone, X10, Ncast, …  Collaboration Voicemail, discussion forum,… Programmable SIP proxy Multimedia application components Programmable IP telephony services 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:wenyu@cs Telephone network SIP/PSTN gateway SIP server (sipd) IP endpoint Telephone subscriber +1 212 9397040 sip:7141@cs.columbia.edu PSTN interworking Load sharing and failover in SIP P P P P P P2P VoIP using SIP SIP-H.323gateway 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. Slave Master Web scripts D2 P2 Master Slave Web scripts 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 Bi-directional replication Program Call routing SIPSAP RSVPRTCP RTP Media G.711 MPEG RTSP Signaling Quality of serviceMedia transport Internet Telephony Internet 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 PA registrar Presence server office.com SUBSCRIBE NOTIFY REGISTER alice@home.com bob@office.com PUA PUA + PA Presence and event notification 1 2 5 2 4 7 6 3 SIP323 SIP/PSTN PSTN phone vxml H.323 clients e.g., NetMeeting High quality Low bitrate gatekeeper sipd SIPH.323 sipd sipum rtspd Alice’s phone Bob’s phone cgi CPL SQL First stage stateless proxy server farm Second stage proxy/registrar (sipd) Libraries (C/C++) SIP, RTP, audio mixing, DB interface, SNMP interface, RTSP, DNS SRV/NAPTR, win32 portability,… Transport layer (TCP/UDP) Client Branch SIP proxy SIPUA API RTSP API RTSP server RTP Interface RTSP tr SIP transaction HTTP Message Parsing