Transcript No Slide Title

“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
©2000, Columbia University
Multimedia Communication
Protocols
Interactive
Internet
Telephony
voice response
Internet
Radio/TV
Messaging
and Presence
Video
conferencing
Unified
messaging
Media
Application layer
Transport (TCP, UDP)
G.711
MPEG
H.323
SIP
RTSP RSVP RTCP
Network (IPv4, IPv6)
RTP
Link layer
Physical layer
Signaling
Quality of
service
Media
transport
CINEMA modules
CINEMA Applications
RTSP media
server
SIP proxy
server
SIP/H.323
gateway
rtspd
sipd
sip323
LDAP
Xerces-C
SIP/RTP
conferencing
SIP/RTSP
SIP/VoiceXML
unified messaging
browser
sipconf
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
Our IP telephony test-bed
Telephone
Telephone
switch
rtspd
RTSP media
server
sipconf
SIP
conference
server
Department
PBX
T1/E1
RTP/SIP
sipd
SIP proxy,
redirect
server
Quicktime
RTSP
RTSP clients
sipum
SIP/RTSP
Unified
messaging
Web based
configuration
Web server
SQL
database
SIP/PSTN Gateway
SNMP
(Network
Management)
e*phone
Device GW
Hardware
Internet (SIP)
phones
X 10
NetMeeting
siph323
SIPH.323
convertor
sipc
Software SIP
user agents
H.323
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
1 Call 9397134
713x is called a part of
Coordinated Dial Plan
(CDP) in a Nortel PBX
Gateway
Internal T1/CAS
(Ext:7130-7139)
2
Call 7134
Ethernet
Regular phone
(internal)
5
3
SIP server
• Direct Inward Dial (DID) direct and simple
• No-DID - dial extension,
supports more users
sipc
Bob’s phone
SQL
database
sipd
4
7134 => bob
IP to PSTN Call
PBX
PSTN
External T1/CAS
5 Call 5551212
Gateway
(10.0.2.3)
Internal T1/CAS
4 Call 85551212
3
Ethernet
5551212
Regular phone
(internal, 7054)
Note: In this direction
there is no distinction
between DID and nonDID calls.
1
Bob calls
5551212
SIP server
sipc
2
SQL
database
sipd
Use sip:[email protected]
Other Applications
RTSP server
SIPUA
API
RTSP transaction
SIP transaction
RTSP API
RTP
Interface
SIP
proxy
Client Branch
HTTP Message Parsing
Transport layer (TCP/UDP)
Layered Libraries
User Interaction
• Web interface
– Administration
– User configuration
• Unified Messaging
– Notify by email
– rtsp or http
• Portal Mode
– 3rd party IpTelSP
http://www.cs.columbia.edu/~kns10/research/cinema
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 IP-Telephony Workshop (IPTel'2000), April 2000.
sipconf
SIP323
sipc
SIP/PSTN
•
•
•
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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
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, 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 based voicemail
Wide range of applicability
Campus/corporate network
rtspd
sipum
Internet
sipum
rtspd
Within a domain
External application
service provider
SipVxml
VoiceXML is a language for specifying voice dialogs for
interactive voice response systems. It is specified in XML.
PSTN
SIP/PSTN gateway
Fetch VoiceXML pages
Web server
CGI, servlet, JSP
Call Request
SIP user agent
Get streaming media
SIP based VoiceXML
browser
SIP phone
Press 1 to listen to next message,
2 to forward …
Media server
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
http://www.sipstone.org
Development Libraries
(User agent API, SIP Stack)
Multiparty Conferencing
Software SIP clients
Programmable SIP
Hardware SIP phones
servers (CGI, CPL)
Services and
applications
Instant messaging
and presence
(In progress)
SIP/H.323 translation
Unified messaging,
voice mail and
answering machine
SIP/VoiceXML browser
(In progress)
Real-time Media Streaming
SIP-PSTN gateway
(In progress)
… moving from IP telephony to
a real-time multimedia collaboration portal…