Transcript No Slide Title

“A flexible architecture to support wide range of multimedia
communication applications, both clients and servers”
©2000, Columbia University
Desktop
SIP clients
H.323
SIPUA
sip323
sipd
Gatekeeper
SIP-H.323
signaling gateway
SIP
Programmable
SIP servers
Conferencing
Hardware SIP phone
sipconf
Unified messaging
sipum
sipgw
SIP-MGCP gateway
rtspd
MGCP
Streaming media
SIP-PSTN gateway
PSTN
Quick-time
Regular telephones
RTSP
Other Applications
RTSP server
SIPUA
API
RTSP transaction
SIP
proxy
RTSP API
RTP
Interface
SIP transaction
Client Branch
HTTP Message Parsing
Transport layer (TCP/UDP)
Architecture overview
sipconf
rtspd
sip323
sipd
sipum
RTSP library
SIP library
HTTP library
RTP/RTCP
HTTP Message Parsing
Transport layer (TCP/UDP)
Example applications based on CINEMA
H.323
Columbia SIP library
Basic and Digest
User registration
CGI/CPL upload
Dynamic session
change
Components
Authentication
Authentication
User registration
SIP/SDP parser
Dynamic session change
SIP/SDP Parser
to be added ...
• Call transfer
• Three party call
• Instant messaging and presence
http://www.cs.columbia.edu/~kns10/software/siplib
CINEMA modules
RTSP media
server
SIP proxy
server
SIP/H.323
gateway
rtspd
sipd
sip323
LDAP
Berkeley DB
xml4j
SIP
conferencing
sipconf
SIP/RTSP
unified messaging
SIP/MGCP
gateway
sipum
sipgw
OpenH323
CINEMA
PGP
PWLib
Resparse
libNT
libcine
libsip
libsip++
Win32
stub
Utilities
parsing
Basic
SIP
library
SIP UA
library
libmixer
RTP
audio
mixer
libdict
Hash
table
libdb++
mySQL
intf
SIP
sipc
H.323
Gatekeeper
• 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
K. Singh, H.Schulzrinne, "Interworking Between SIP/SDP and H.323". Proceedings of the 1st IPTelephony Workshop (IPTel'2000), April 2000.
sipconf
SIP323
sipc
SIP/PSTN
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•
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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
http://www.cs.columbia.edu/~kns10/software/sipconf
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.
SIP/RTSP based unified messaging
Wide range of applicability
Campus/corporate network
Internet
rtspd
sipum
sipum
rtspd
Within a domain
External application
service provider
Development Libraries
(User agent API, SIP Stack)
Multiparty Conferencing
Software SIP clients
Programmable SIP
servers (CGI, CPL)
Hardware SIP phones
Services and
applications
Instant messaging
and presence
(In progress)
Unified messaging,
voice mail and
answering machine
Web to phone
SIP/H.323 translation
(In progress)
Real-time Media Streaming
SIP-PSTN gateway
(In progress)