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sipd sip323 sipconf sipum sipvxmlrtspd
CINEMA Libraries
libNT
Win32 stub
libcine
Utilities parsingIPv6
libsip
Basic SIP library
libsip++
SIP UA library
libmixer
RTP audio mixer
libdict
Hash table
libdb++
mySQL intf
RTSP mediaserver
SIP proxy server
SIP/H.323gateway
SIP/RTP conferencing
SIP/RTSP unified messaging
SIP/VoiceXMLbrowser
LDAPXerces-C OpenH323
MySQLPWLibResparse
librtsp
RTSPclient
librtp
RTP library
libsnmp
SIP MIB
ViaVoiceXerces-C
CINEMA Applications
““A flexible architecture to support wide range of multimedia A flexible architecture to support wide range of multimedia communication applications, both clients and servers”communication applications, both clients and servers”
http://www.cs.columbia.edu/IRT/cinema/http://www.cs.columbia.edu/IRT/cinema/
Presented by:
Kundan SinghJoint work with Wenyu Jiang, Jonathan Lennox, Sankaran Narayanan, Henning Schulzrinne and Ali Khwaja
Telephone7040
SIP/PSTN Gateway
Department PBX
Web based configuration
Web server
Telephoneswitch
Device GW
X 10
SQLdatabase
sipd
7134,wenyu
Xiaotaow
NetMeetingsiph323
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,ArchitectureArchitecture
• 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
SIPSIP H.323H.323
Gatekeepersipc
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
sipcsipcSIP323SIP323
SIP/PSTNSIP/PSTN
K. Singh, G.Nair and H.Schulzrinne, “Centralized Conferencing using SIP". Proceedings of the 2st IP-Telephony Workshop (IPTel'2001), April 2001.
Multimedia ConferencingMultimedia Conferencing
Unified MessagingUnified Messagingvoice 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 agentSIP user agent
SIP/PSTN gatewaySIP/PSTN gateway
Web serverWeb server
CGI, servlet, JSP
SIP based VoiceXML
browser
SIP phoneSIP phone
Media serverMedia 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.
Performance Performance measurement measurement
and Scalabilityand 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.orghttp://www.sipstone.org
Services and Services and applicationsapplications
Multiparty ConferencingMultiparty Conferencing
Unified messaging,Unified messaging,voice mail and voice mail and answering machineanswering machine
SIP/VoiceXML browserSIP/VoiceXML browser(In progress)(In progress)
Real-time Media StreamingReal-time Media StreamingSIP/H.323 translationSIP/H.323 translation
Hardware SIP phonesHardware SIP phones
Instant messagingInstant messagingand presenceand presence(In progress)(In progress)
SIP-PSTN gatewaySIP-PSTN gateway(In progress)(In progress)
Software SIP Software SIP clientsclients
Development Libraries Development Libraries (User agent API, SIP (User agent API, SIP
Stack)Stack)
Programmable Programmable SIP servers SIP servers (CGI, CPL)(CGI, CPL)
… moving from IP telephony to a real-time multimedia collaboration environment…