1. SIP-based VoiceXML browser (sipvxml)
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
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
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 P
P2P VoIP using SIP
First stage stateless proxy server farm
Unified messaging using SIP and RTSP M
rtspd: media server
Local/long distance
e.g.,
sipconf:
conference server a1
a.example.com
_sip._udp
SRV 0 0 a1
SRV 1 0 a2
Telephone
switch 1 2 5 4 7 6 3
RTSP S
RTSP clients
e.g., Quicktime s1 a2
PSTN
Department
PBX
sipd
Bob’s 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
Internal
Telephone
e.g., 7040
sipd:
proxy,
redirect,
registrar
sipum s2
Alice’s phone
713x M
SQL
database
cgi
Web server s3 b1
b.example.com
_sip._udp
SRV 0 0 b1
SRV 1 0 b2
Alice (caller) calls Bob
The SIP server forks the call to Bob’s phone and the mail server
After 10 seconds, the mail server sets up RTSP sessions to playback welcome message and to record mail
Mail server accepts the call
SIP server cancels the other branch
SIP server forwards the acceptance
Media packets are sent directly between the RTSP server and caller
vxml S
SIP/PSTN Gateway
e.g., Cisco 2600
Web based
configuration b2
rtspd
SIP
VXML
Presence and event notification
Web
scripts
Web
scripts
7134
7136
H.323
office.com D1 D2
siph323:
SIP-H.323
translator
Master
Slave
Presence server
Bi-directional
replication
Slave
Master
(software phone) PA
PUA
H.323 clients
e.g., NetMeeting
REGISTER P1 P2
SUBSCRIBE
Multimedia conferencing
NOTIFY
PUA
registrar
gatekeeper
SIP-H.323
gateway
phone.cs.columbia.edu
sip2.cs.columbia.edu
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).
PUA + PA
REGISTER
Low bitrate
SIP323
SIP
H.323
_sip._udp
SRV phone.cs.columbia.edu
SRV sip2.cs.columbia.edu
High quality
proxy1 = phone.cs
backup = sip2.cs
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
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
Call request
Web server
RTSP
CGI, servlet, JSP
RTP
Other Applications
Application layer
SIP phone
            
Get streaming media
Transport (TCP, UDP)
SIP-based VoiceXML browser (sipvxml)
Libraries (C/C++)
SIP, RTP, audio mixing, DB interface, SNMP interface, RTSP, DNS SRV/NAPTR, win32 portability,…
Transport layer (TCP/UDP)
RTP
Interface
HTTP Message Parsing
RTSP transaction
SIP transaction
Client Branch
RTSP API
RTSP server
SIPUA
API
SIP
proxy
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
Physical layer
Program
Call
routing
Voice
XML
DTMF
Mixing
Speech/
text
SDP
PSTN interworking
IP endpoint
… moving from IP telephony to real-time multimedia collaboration…
Telephone
network
Telephone
subscriber
SIP/PSTN
gateway
SIP server
(sipd)
Layered Architecture