©2000, Columbia University “A flexible architecture to support wide range of multimedia communication applications, both clients and servers” Presented by: Kundan Singh Joint work with Wenyu Jiang, Jonathan Lennox, Sankaran Narayanan, Henning Schulzrinne and Xiaotao Wu at Columbia University

Physical layer Link layer Network (IPv4, IPv6) Transport (TCP, UDP) Application layer H.323RTSPRSVPRTCP RTP Media G.711 MPEG SIP Signaling Quality of service Media transport Internet Telephony Internet Radio/TV Messaging and Presence Interactive voice response Unified messaging Video conferencing Multimedia Communication Protocols

CINEMA modules 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

e*phone sipc Software SIP user agents Hardware Internet (SIP) phones Our IP telephony test-bed SIP proxy, redirect server SQL database sipd T1/E1 RTP/SIP Telephone SIP/PSTN Gateway Department PBX Web based configuration Web server Telephone switch SNMP (Network Management) SIPH.323 convertor NetMeeting siph323 H.323 rtspd SIP/RTSP Unified messaging RTSP media server sipum Quicktime RTSP clients RTSP SIP conference server sipconf Device GW X 10 W. Jiang, J. Lennox, H. Schulzrinne and K. Singh, “Towards Junking the PBX: Deploying IP Telephony". NOSSDAV 2001,

PSTN to IP Call PBX PSTN External T1/CAS Regular phone (internal) Call SIP server sipd Ethernet 3 SQL database => bob sipc 5 Bob’s phone Direct Inward Dial (DID) - direct and simple No-DID - dial extension, supports more users Gateway Internal T1/CAS (Ext: ) Call x is called a part of Coordinated Dial Plan (CDP) in a Nortel PBX

IP to PSTN Call Gateway ( ) 3 SQL database 2 Use Ethernet SIP server sipd sipc 1 Bob calls PSTN External T1/CAS Call PBX Internal T1/CAS Call Regular phone (internal, 7054) Note: In this direction there is no distinction between DID and non- DID calls.

Layered Libraries 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

User Interaction Web interface –Administration –User configuration Unified Messaging –Notify by –rtsp or http Portal Mode –3 rd party IpTelSP

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 K. Singh, G.Nair and H.Schulzrinne, “Centralized Conferencing using SIP". Proceedings of the 2st IP-Telephony Workshop (IPTel'2001), April 2001.

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 based voic SIP based voic Wide range of applicability Campus/corporate network sipum rtspd Within a domain Internet sipum rtspd External application service provider

VoiceXML is a language for specifying voice dialogs for interactive voice response systems. It is specified in XML. SipVxml 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 …

Performance measurement and Scalability Busy hour call arrival (BHCA) Requests per second (proxy) Request turn-around time (proxy) Participants per conference (sipconf) Simultaneous media streams (rtspd) DNS based scalability with server farms Stateless proxy Hierarchical conference servers Redirect feature

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 portal…