1. ©2000, Columbia University “A flexible architecture to support wide range of multimedia communication applications, both clients and servers”

2. rtspd Quick-time Gatekeeper SIPUA SIP H.323 RTSP sipd sipconf sipum sip323 SIP-H.323 signaling gateway Conferencing Programmable SIP servers Unified messaging Streaming media Hardware SIP phone Desktop SIP clients sipgw PSTN MGCP SIP-MGCP gateway SIP-PSTN gateway Regular telephones

3. Architecture overview 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

4. Example applications based on CINEMA Transport layer (TCP/UDP) RTP/RTCP HTTP Message Parsing RTSP library SIP library HTTP library H.323 sip323 sipum sipd sipconf rtspd

5. SIP/SDP Parser Authentication User registration Dynamic session change SIP/SDP parser Authentication Basic and Digest User registration CGI/CPL upload Dynamic session change Components to be added... Call transfer Three party call Instant messaging and presence Easy to use ! Columbia SIP library http://www.cs.columbia.edu/~kns10/software/siplib

6. CINEMA modules sipdsip323sipconfsipumsipgwrtspd CINEMA libNT Win32 stub libcine Utilities parsing 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 conferencing SIP/RTSP unified messaging SIP/MGCP gateway LDAP Berkeley DB xml4j OpenH323 PGP PWLib Resparse

7. 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 http://www.cs.columbia.edu/~kns10/software/sip323 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.

8. sipconf sipc http://www.cs.columbia.edu/~kns10/software/sipconf SIP323 SIP/PSTN 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

9. SIP/RTSP based unified messaging voice mail, answering machine, web based setup, email and web integration... http://www.cs.columbia.edu/~kns10/software/sipum Kundan Singh and Henning Schulzrinne, "Unified Messaging using SIP and RTSP". IP Telecom Services Workshop 2000, Sept 2000. Atlanta, Georgia.

10. SIP/RTSP based unified messaging SIP/RTSP based unified messaging Wide range of applicability Campus/corporate network sipum rtspd Internet sipum Within a domain External application service provider

11. Services and applications Multiparty Conferencing Unified messaging, voice mail and answering machine Web to phone (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)