CINEMA (Columbia InterNet Extensible Multimedia Architecture) presented by

1
CINEMA (Columbia InterNet Extensible Multimedia
Architecture)presented by Kundan Singh, Joint
work with Wenyu Jiang, Jonathan Lennox, Sankaran
Narayanan, Henning Schulzrinne, Xiaotao WuMore
information at http//www.cs.columbia.edu/IRT/cine
ma/

• Project Objectives
• 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.

• Performance
• 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

• Approach
• Develop protocols (SIP, RTSP, RTP,)
• Implement common reusable libraries
• Provide distributed servers components
• Integrate with web, email, phone systems

Session Initiation Protocol (SIP)-based
enterprise VoIP infrastructure
Load sharing and failover in SIP
example.com _sip._udp SRV 0 0 s1 SRV 0 0 s2
SRV 0 0 s3 SRV 1 0 ex
Second stage proxy/registrar (sipd)
CINEMA servers
P2P VoIP using SIP
First stage stateless proxy server farm
Unified messaging using SIP and RTSP
rtspd media server
a1
a.example.com _sip._udp SRV 0 0 a1 SRV 1 0 a2
sipconf conference server
Local/long distance e.g., 1-212-5551212
Telephone switch
RTSP
s1
a2
RTSP clients e.g., Quicktime
PSTN
Department PBX
sipd
Bobs phone
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 unified messaging
sipbob_at_example.com
sipum
sipbob_at_b.example.com
sipd proxy, redirect, registrar
Internal Telephone e.g., 7040
s2
Alices phone
713x
SQL database
b.example.com _sip._udp SRV 0 0 b1 SRV 1 0 b2
cgi
s3
b1

1. Alice (caller) calls Bob
2. The SIP server forks the call to Bobs 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

vxml
SIP/PSTN Gateway e.g., Cisco 2600
Web based configuration
b2
rtspd
Presence and event notification
Web scripts
Web scripts
7134
D2
D1
office.com
H.323
7136
bob_at_office.com
siph323 SIP-H.323 translator
Slave Master
Master Slave
Bi-directional replication
Presence server
alice_at_cs.columbia.edu (software phone)
PUA
PA
H.323 clients e.g., NetMeeting
alice_at_home.com
REGISTER
P2
P1
SUBSCRIBE
Multimedia conferencing
NOTIFY
PUA
registrar
gatekeeper
phone.cs.columbia.edu
sip2.cs.columbia.edu
SIP-H.323 gateway
sipd
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 play-out 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).
REGISTER
PUA PA
Low bitrate
SIP323
SIP
H.323
_sip._udp SRV 0 0 5060 phone.cs.columbia.edu
SRV 1 0 5060 sip2.cs.columbia.edu
proxy1 phone.cs backup sip2.cs
High quality
A signaling translator between ITU-Ts multistage
H.323 and IETFs SIP that supports different
dialing modes, has a built-in gatekeeper and is
transparent to media path.

• Overview
• Multimedia communication
• Audio, video, text, screen sharing,
• PSTN interworking, IVR
• Multi-devices
• IP-phone, telephone, X10, Ncast,
• Collaboration
• Voicemail, discussion forum,

Multimedia application components
sipconf
Interactive voice response
SIP/PSTN
Internet Telephony
Internet Radio/TV
Programmable SIP proxy
Messaging and Presence
Programmable IP telephony services
Unified messaging
Interactive voice response (IVR)
Video conferencing
PSTN phone
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
SIP/PSTN gateway
vxml
cgi CPL
SIP
SAP
RSVP
RTCP
Media G.711 MPEG
SQL
Fetch VoiceXML pages
Web server
Call request
RTSP
CGI, servlet, JSP
RTP
Application layer
Other Applications
SIP phone
Get streaming media
Transport (TCP, UDP)
Libraries (C/C) SIP, RTP, audio mixing, DB
interface, SNMP interface, RTSP, DNS SRV/NAPTR,
win32 portability,
SIP-based VoiceXML browser (sipvxml)
Quality of service
Media transport
Network (IPv4, IPv6)
Signaling
Press 1 to listen to next message, 2 to forward
Media server
Link layer
SIP phone
Program Call routing
Voice XML
DTMF
Mixing
Speech/ text
SDP
Physical layer
PSTN interworking
IP endpoint
1 212 9397040
moving from IP telephony to
real-time multimedia collaboration
sipwenyu_at_cs
Telephone network
Telephone subscriber
SIP/PSTN gateway
Layered Architecture
SIP server (sipd)
sip7141_at_cs.columbia.edu