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

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Presentation transcript:

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

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

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

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

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

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

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.

sipconf sipc 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

SIP/RTSP based unified messaging voice mail, answering machine, web based setup, and web integration... Kundan Singh and Henning Schulzrinne, "Unified Messaging using SIP and RTSP". IP Telecom Services Workshop 2000, Sept Atlanta, Georgia.

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

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)