CINEMA (Columbia InterNet Extensible Multimedia Architecture)
presented by – Kundan Singh, Joint work with Wenyu Jiang, Jonathan Lennox, Sankaran Narayanan, Henning Schulzrinne, Xiaotao Wu
More information at http://www.cs.columbia.edu/IRT/cinema/
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Project Objectives
Approach
Develop protocols (SIP, RTSP, RTP,…)
Implement common reusable libraries
Provide distributed servers components
Integrate with web, email, phone systems
Performance
A flexible architecture to support clients and servers for wide
range of multimedia communication applications such as video
conferencing, Internet telephony/radio, interactive voice
response, unified messaging, presence and multimedia
collaboration.
Session Initiation Protocol (SIP)-based
enterprise VoIP infrastructure
Load sharing and failover in SIP
P
CINEMA servers
Local/long distance
e.g., 1-212-5551212
sipconf:
conference server
Telephone
switch
PSTN
RTSP
RTSP clients
e.g., Quicktime
SQL
database
cgi
Web
server
vxml
SIP/PSTN Gateway
e.g., Cisco 2600
Web based
configuration
VXML
7134
H.323
siph323:
SIP-H.323
translator
alice@cs.columbia.edu
(software phone)
Bob’s phone
2
7
4
3
High
quality
Web server
CGI, servlet, JSP
Call request
SIP phone
SIP-based VoiceXML
browser (sipvxml)
SIP phone
+1 212 9397040
Press 1 to listen to next message,
2 to forward …
PSTN interworking
Telephone
subscriber
Telephone
network
SIP/PSTN
gateway
SIP server
(sipd)
Bi-directional
replication
Slave
Master
P2
PUA
sip2.cs.columbia.edu
REGISTER
_sip._udp
SRV 0 0 5060 phone.cs.columbia.edu
SRV 1 0 5060 sip2.cs.columbia.edu
Overview
 Multimedia communication
Internet
Telephony
Interactive
voice response
Internet
Radio/TV
Messaging
and Presence
 Multi-devices
 Collaboration
proxy1 = phone.cs
backup = sip2.cs
Multimedia application components
Audio, video, text, screen sharing, …
PSTN interworking, IVR
Unified
messaging
Video
conferencing
Voicemail, discussion forum,…
SQL
Media
SIP
RTSP
G.711
MPEG
SAP
RSVP
RTCP
Application layer
RTP
Other Applications
Libraries (C/C++)
Transport (TCP, UDP)
Network (IPv4, IPv6)
RTSP API
RTP
Interface
Quality of service
Signaling
Media transport
Link layer
RTSP server
RTSP tr
SIPUA
API
SIP
proxy
Physical layer
Program
Call
routing
Voice
XML
DTMF
Mixing
Speech/
text
SDP
SIP transaction
HTTP Message Parsing
Client Branch
Transport layer (TCP/UDP)
sip:7141@cs.columbia.edu
S
P1
IP-phone, telephone, X10, Ncast, …
SIP, RTP, audio mixing, DB
interface, SNMP interface,
RTSP, DNS SRV/NAPTR,
win32 portability,…
sip:wenyu@cs
b2
b.example.com
_sip._udp
SRV 0 0 b1
SRV 1 0 b2
D2
Master
Slave
REGISTER
H.323
Media server
IP endpoint
M
Web
scripts
PUA + PA
Programmable SIP proxy
cgi
CPL
b1
phone.cs.columbia.edu
A signaling translator between ITU-T’s multistage H.323
and IETF’s SIP that supports different dialing modes, has
a built-in gatekeeper and is transparent to media path.
Programmable call routing based
on time of day, caller id, etc., using
server side Call processing
language, Common Gateway
interface (CPL), Java servlets or
client side Language for End
System services (LESS) scripts
S
a.example.com
_sip._udp
SRV 0 0 a1
SRV 1 0 a2
D1
gatekeeper
SIP
a2
Web
scripts
PUA
registrar
sipd
M
sip:bob@b.example.com
s3
SUBSCRIBE
Programmable IP
telephony services
vxml
s2
Presence server
alice@home.com
SIP-H.323
gateway
a1
sip:bob@example.com
bob@office.com
NOTIFY
A SIP/RTP-based centralized conference server
to support audio mixing, video forwarding, text
chat and screen sharing among heterogeneous
endpoints such as PC and phones. It has playout delay adjustment for wide area Internet,
web-based conference setup, high quality
audio (G.722, G.711) as well as low bit rate
codecs (GSM, DVI).
s1
P
PA
Interactive voice
response (IVR)
SIP/PSTN gateway
P
office.com
SIP/PSTN
PSTN phone
P
First stage stateless
proxy server farm
Presence and event notification
H.323 clients
e.g., NetMeeting
Low
bitrate
P
Second stage proxy/registrar (sipd)
rtspd
Multimedia conferencing
SIP323
example.com
_sip._udp
SRV 0 0 s1
SRV 0 0 s2
SRV 0 0 s3
SRV 1 0 ex
Peer-to-peer Internet telephony avoids the
configuration and maintenance cost of
server-based architecture and dependency
on controlled infrastructure such as DNS. We
use Chord algorithm on top of SIP for an
interoperable, scalable and robust P2P-SIP
endpoint.
sipum
1. Alice (caller) calls Bob
2. The SIP server forks the call to Bob’s phone
and the mail server
3. After 10 seconds, the mail server sets up
RTSP sessions to playback welcome
message and to record mail
4. Mail server accepts the call
5. SIP server cancels the other branch
6. SIP server forwards the acceptance
7. Media packets are sent directly between the
RTSP server and caller
SIP
7136
5
6
Alice’s phone
P2P VoIP
using SIP
2
sipd
1
sipum:
unified
messaging
sipd:
proxy,
redirect,
registrar
713x
Unified messaging using SIP
and RTSP
rtspd: media server
Department
PBX
Internal
Telephone
e.g., 7040
sipstone: benchmark for SIP servers
Different signaling vs. media components
Black-box measurement and white-box profiling
Load balancing, thread pooling, and reactive
system to improve performance
 Novel peer-to-peer IP telephony using SIP
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Layered Architecture
… moving from IP telephony to
real-time multimedia collaboration…