1. sipdsip323sipconfsipumsipvxmlrtspd CINEMA Libraries libNT Win32 stub libcine Utilities parsing IPv6 libsip Basic SIP library libsip++ SIP UA library libmixer RTP audio mixer libdict Hash table libdb++ mySQL intf RTSP media server SIP proxy server SIP/H.323 gateway SIP/RTP conferencing SIP/RTSP unified messaging SIP/VoiceXML browser LDAP Xerces-C OpenH323 MySQL PWLib Resparse librtsp RTSP client librtp RTP library libsnmp SIP MIB ViaVoice Xerces-C CINEMA Applications “A flexible architecture to support wide range of multimedia communication applications, both clients and servers” http://www.cs.columbia.edu/IRT/cinema/ Presented by : Kundan Singh Joint work with Wenyu Jiang, Jonathan Lennox, Sankaran Narayanan, Henning Schulzrinne and Ali Khwaja

2. Telephone 7040 SIP/PSTN Gateway Department PBX Web based configuration Web server Telephone switch Device GW X 10 SQL database sipd 7134,wenyu Xiaotaow NetMeeting siph323 H.323 rtspd sipum Quicktime RTSP clients RTSP sipconf 7135, sank 713x Single Box (Netra) Ncast video encoder SNMP (Network Management) W. Jiang, J. Lennox, H. Schulzrinne and K. Singh, “Towards Junking the PBX: Deploying IP Telephony". NOSSDAV 2001, Architecture

3. Inter-working between SIP and H.323 version 2.0 H.323 fast-start as well as normal call Multiple simultaneous independent calls Transparent media traffic Unix as well as Windows Built-in gatekeeper Different dialing modes SIPH.323 Gatekeeper sipc K. Singh, H.Schulzrinne, "Interworking Between SIP/SDP and H.323". Proceedings of the 1st IP- Telephony Workshop (IPTel'2000), April 2000. SIP based conferencing server SIP/SDP and RTP/RTCP Audio mixing Play-out delay algorithm Web based conference setup G.711 A and Mu law, G.721, DVI ADPCM Multiple simultaneous conferences sipc SIP323 SIP/PSTN K. Singh, G.Nair and H.Schulzrinne, “Centralized Conferencing using SIP". Proceedings of the 2st IP-Telephony Workshop (IPTel'2001), April 2001. Multimedia Conferencing

4. Unified Messaging voice mail, answering machine, web based setup, email and web integration... Kundan Singh and Henning Schulzrinne, "Unified Messaging using SIP and RTSP". IP Telecom Services Workshop 2000, Sept 2000. Atlanta, Georgia. 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 … VoiceXML is an XML based language for specifying voice dialogs for interactive voice response systems.

5. Performance measurement and Scalability Busy hour call arrival (BHCA) Requests per second Request turn-around time Participants per conference Simultaneous media streams DNS based scalability with server farms Stateless proxy Hierarchical conference servers Redirect feature http://www.sipstone.org Services and applications Multiparty Conferencing Unified messaging, voice mail and answering machine SIP/VoiceXML browser (In progress) Real-time Media Streaming SIP/H.323 translation Hardware SIP phones Instant messaging and presence (In progress) SIP-PSTN gateway (In progress) Software SIP clients Development Libraries (User agent API, SIP Stack) Programmable SIP servers (CGI, CPL) … moving from IP telephony to a real-time multimedia collaboration environment…