Transcript No Slide Title

Presented by:
Kundan Singh
Joint work with Wenyu
Jiang, Jonathan Lennox,
“A flexible architecture to support wide range of multimedia Sankaran Narayanan,
Henning Schulzrinne and
communication applications, both clients and servers”
Ali Khwaja
CINEMA Applications
RTSP media
server
rtspd
SIP proxy
server
sipd
SIP/H.323
gateway
sip323
LDAP
Xerces-C
SIP/RTP
conferencing
sipconf
SIP/RTSP
SIP/VoiceXML
unified messaging
browser
sipum
sipvxml
ViaVoice
Xerces-C
OpenH323
CINEMA Libraries
libNT
libcine
libsip
Win32
stub
Utilities
parsing
IPv6
Basic
SIP
library
MySQL
PWLib
Resparse
librtsp
RTSP
client
libsip++
SIP UA
library
librtp
RTP
library
libmixer
RTP
audio
mixer
libdict
libdb++
libsnmp
Hash
table
mySQL
intf
SIP
MIB
http://www.cs.columbia.edu/IRT/cinema/
Architecture
W. Jiang, J. Lennox, H. Schulzrinne and K. Singh, “Towards
Junking the PBX: Deploying IP Telephony". NOSSDAV 2001,
Telephone
switch
Single Box
(Netra)
Telephone
7040
rtspd
Quicktime
RTSP
sipconf
RTSP clients
Web based
configuration
sipum
Department
PBX
713x
Web server
sipd
SQL
database
Ncast video
encoder
SIP/PSTN Gateway
SNMP
(Network
Management)
Device GW
X 10
7134,wenyu
NetMeeting
7135, sank
siph323
H.323
Xiaotaow
Multimedia Conferencing
K. Singh, G.Nair and H.Schulzrinne, “Centralized Conferencing using SIP".
Proceedings of the 2st IP-Telephony Workshop (IPTel'2001), April 2001.
SIP323
sipc
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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/PSTN
SIP
H.323
• 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
sipc
Gatekeeper
K. Singh, H.Schulzrinne, "Interworking Between
SIP/SDP and H.323". Proceedings of the 1st IPTelephony Workshop (IPTel'2000), April 2000.
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.
SIP user agent
SIP phone
PSTN
SIP/PSTN gateway
Call Request
Press 1 to
listen to next
message,
2 to forward
…
SIP based
VoiceXML
browser
Fetch
VoiceXML
pages
Web server
CGI, servlet, JSP
Get
streaming
media
Media server
VoiceXML is an XML based language for
specifying voice dialogs for interactive voice
response systems.
Development Libraries
(User agent API, SIP
Stack)
Performance
measurement
and Scalability
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Software SIP
Multiparty Conferencing
Busy hour call arrival (BHCA)
clients
Requests per second
Request turn-around time
Programmable
Participants per conference
SIP servers
Hardware SIP phones
Simultaneous media streams
(CGI, CPL)
DNS based scalability with
Services and
server farms
Unified messaging,
applications
Stateless proxy
voice mail and
Hierarchical conference
answering machine
Instant messaging
servers
and presence
Redirect feature
(In progress)
SIP/VoiceXML browser
http://www.sipstone.org
(In progress)
SIP/H.323 translation
Real-time Media Streaming
SIP-PSTN gateway
(In progress)
… moving from IP telephony to
a real-time multimedia collaboration environment…