1. SIP based VoiceXML browser
CINEMA (Columbia InterNet Extensible Multimedia Architecture) presented by – Kundan Singh, Joint work with Wenyu Jiang, Jonathan Lennox, Sankaran Narayanan, Henning Schulzrinne, Xiaotao Wu Columbia University, New York. More information at
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, , phone systems
Session Initiation Protocol (SIP)-based enterprise VoIP infrastructure
Load sharing and failover in SIP
Internal
Telephone
Extn: 7040
SIP/PSTN Gateway
Department
PBX
Web based
configuration
Web server
switch
SQL
database
sipd:
Proxy, redirect,
Registrar server
NetMeeting
H.323
rtspd: media server
sipum:
Unified
messaging
Quicktime
RTSP clients
RTSP
713x
CINEMA servers
sipconf:
Conference server
siph323:
SIP-H.323
translator
Local/long distance
PSTN
SIP
VXML
vxml
cgi
example.com
_sip._udp
SRV 0 0 s1
SRV 0 0 s2
SRV 0 0 s3
SRV 1 0 ex M S s1 s2 s3 a1 a2 b1 b2
a.example.com
SRV 0 0 a1
SRV 1 0 a2
b.example.com
SRV 0 0 b1
SRV 1 0 b2 P
P2P VoIP using SIP
Unified messaging using SIP and RTSP
Peer-to-peer Internet telephony avoids the configuration and maintenance cost of server-based infrastructure 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.
Master
Web
scripts D2 P2 D1 P1
phone.cs.columbia.edu
sip2.cs.columbia.edu
REGISTER
proxy1 = phone.cs
backup = sip2.cs
_sip._udp
SRV phone.cs.columbia.edu
SRV sip2.cs.columbia.edu
replication
Presence and event notification PA
registrar
Presence server
office.com
SUBSCRIBE
NOTIFY
REGISTER
PUA
PUA + PA
Multimedia conferencing
sipc
sipconf
SIP323
Netmeeting
SIP/PSTN
SIP
H.323
Netmeeting
e*phone
sipd
sip323
SIP-H.323
gateway
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).
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.
Overview
Multimedia communication
Audio, video, text, screen sharing, …
PSTN interworking, IVR
Multi-devices
IP-phone, telephone, X10, Ncast, …
Collaboration
Voic , discussion forum,…
Multimedia application components
Program
Call
routing
SIP
SAP
RSVP
RTCP
RTP
Media
G.711
MPEG
RTSP
Signaling
Quality of service
Media transport
Internet
Telephony
Radio/TV
Messaging
and Presence
Interactive
voice response
Unified
messaging
Video
conferencing
Physical layer
Link layer
Network (IPv4, IPv6)
Transport (TCP, UDP)
Application layer
Voice
XML
DTMF
Mixing
Speech/
text
SDP
SIP “forking”
proxy
Programmable IP
telephony services
PSTN
SIP user agent
SIP/PSTN gateway
Web server
CGI, servlet, JSP
SIP based VoiceXML browser
SIP phone
            
Media server
Call Request
Fetch VoiceXML pages
Get streaming media
Press 1 to listen to next message,
2 to forward …
Interactive voice response
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
Transport layer (TCP/UDP)
RTP
Interface
HTTP Message Parsing
RTSP transaction
SIP transaction
Client Branch
RTSP API
RTSP server
SIPUA
API
SIP
proxy
Other Applications
Libraries (C/C++)
SIP, RTP, audio mixing, DB interface, SNMP interface, RTSP, DNS SRV, win32 portability,…
PSTN interworking
Telephone
network
SIP/PSTN
gateway
SIP server
IP endpoint
subscriber
… moving from IP telephony to
real-time multimedia collaboration…
Layered Architecture