CINEMA – The Columbia CINEMA – The Columbia Internet Multimedia Internet Multimedia ArchitectureArchitecture

Henning SchulzrinneIRT Lab, Dept. of Computer

ScienceNovember 2001

OverviewOverview

Motivation Migration strategy CINEMA architecture and goals Demo Future plans

MotivationMotivation Allow migration of enterprises to IP

multimedia communication Add capacity to existing PBX,

without upgrade Allow both

IP centrex: hosted by carrier “PBX”-style: locally hosted Unlike classical centrex, transition can

be done transparently

Migration strategyMigration strategy

1. Add IP phones to existing PBX or Centrex system – PBX as gateway

2. Add multimedia capabilities: PCs, dedicated video servers

3. “Reverse” PBX: replace PSTN connection with SIP/IP connection to carrier

4. Retire PSTN phones

CINEMACINEMA Integrated communications environment

multimedia: audio, video, shared applications, chat, …

call handling and routing conferencing unified messaging control of networked devices instant messaging and presence

Carrier (hosted) or enterprise environment Integrated with existing PSTN environment

CINEMA – SIP-basedCINEMA – SIP-based SIP = Session Initiation Protocol, but

more … SIP as common infrastructure:

IP telephony signaling (also for 3G wireless) Multimedia conference setup Instant messaging and presence (new AOL

interworking standard) Event notification

Standardized by IETF (RFC 2543) Lots of gateways and hardware phones Library part of Windows XP

CINEMA platformCINEMA platform

Linux Solaris Windows NTWindows 98

FreeBSDOpenBSD

Compaq Tru64

Windows 2000Windows XP

Same code base on all major client and server platforms:

CINEMA componentsCINEMA components

RTSP

sipum

Cisco 7960

sipvxmlSIP

rtspdsipconfLDAP server

MySQL

PhoneJack interface

sipc

T1T1

sipd

mediaserver

RTSP

SIP-H.323converter

messagingserver

unified

server(MCU)

user database

conferencing

sip-h323

VoiceXMLserver

proxy/redirect server

Cisco2600

Pingtel

wireless802.11b

PBX

MeridianNortel

plug'n'sip

sipd – SIP proxy and sipd – SIP proxy and redirect serverredirect server Platform for application

development: SIP cgi script CPL (in progress) servlets (in progress) policy C API

Highly scalable Logging to RADIUS and SQL Tested in several SIP interop tests

sipcsipc

SIP user client Multimedia, with plug-ins for

different media Currently, audio, video, shared

web browsing, application sharing, chat

Instant messaging and presence Integrated device controller

CINEMA software structureCINEMA software structure

sipd sip323 sipconf sipum sipvxmlrtspd

CINEMA Libraries

libNT

Win32 stub

libcine

Utilities parsingIPv6

libsip

Basic SIP library

libsip++

SIP UA library

libmixer

RTP audio mixer

libdict

Hash table

libdb++

mySQL intf

RTSP mediaserver

SIP proxy server

SIP/H.323gateway

SIP/RTP conferencing

SIP/RTSP unified messaging

SIP/VoiceXMLbrowser

LDAPXerces-C

OpenH323

MySQLPWLibResparse

librtsp

RTSPclient

librtp

RTP library

libsnmp

SIP MIB

ViaVoiceXerces-C

CINEMA Applications

DemoDemo IP-PSTN calls via Nortel PBX Web-based call routing SIP-H.323 protocol translation Three-way conference with video Programmable call routing Using SIP in virtual worlds Instant messaging and presence Device control VoiceXML

7040

SIP/PSTN Gateway

Department PBX

Web based configuration

Web server

Telephoneswitch

Device GW

X 10

SQLdatabase

sipd

7134,wenyu

Xiaotaow

NetMeetingsiph323

H.323

rtspd

sipum

Quicktime

RTSP clients

RTSP

sipconf

7136, my163

713x

(Netra)

Ncast video encoder

IP-PSTN callsIP-PSTN calls

Telephone7040

7134,wenyu

SIP/PSTN Gateway

SQLdatabase

sipd713x

Web-based call routingWeb-based call routing

Contacts can be added from web page

SIP phones register automatically Forking: any number of phones can

ring simultaneously or sequentially (with timer)

First to pick up gets the call

SIP-H.323 gatewaySIP-H.323 gateway

SQLdatabase

sipd

7134,wenyuNetMeeting

siph323

H.323

Multiparty conferenceMultiparty conferenceTelephone7040

SIP/PSTN Gateway

Department PBX

SQLdatabase

sipd

7134,wenyu

Xiaotaow

NetMeetingsiph323

H.323

sipconf

7136, my163

713x

Ncast video encoder

Programmable call routingProgrammable call routing

sipd

7134,wenyu

Xiaotaow

rtspd

sipum

Quicktime

RTSP clients

RTSP

High Priority

Low Priority

My owner’s SIP address is sip:john@cs.columbia.edu

sip:john@cs.columbia.edu

Help!!! (invoke sipc to callsip:john@cs.columbia.edu

INVITE sip:john@cs.columbia.edu

Make MM call

Send message

Save message history

ConfigurationTerminate

kns10@bourbon.cs.columbia.edu xiaotaow@cs.columbia.edu

Instant messagingInstant messaging

Do sip:lamp@cs.columbia.edu SIP/2.0…..<Control><Action>turn lamp on</Action></Control>

serial port

Device controlDevice control

VoiceXMLVoiceXML

XML-based scripting language for interactive voice response services

Text-to-speech or sound samples Program like web forms We’ll use it for conferencing, auto-

attendant and UM Initial demo: dial weather

sip.cgi?sip:xyz@cs.columbia.edu

INVITE sip:xyz@cs.columbia.edu

Ncast MPEG distributionNcast MPEG distribution

The future of SIPThe future of SIP

Common infrastructure for interactive services

Adopted for 3G wireless systems To be used in PacketCable

(DOCSIS) IM and presence interworking Emerging: home and industrial

device control

Future plansFuture plans scalable and resilient systems

using commodity computing hardware and software

SIP-VoiceXML server & integration gateway location and call routing Call Processing Language (CPL)

implementation creation and visualization

IPv6 support

Future plans: conferencing Future plans: conferencing

moderated conferences networks of MCUs for load

balancing conference reservations asynchronous collaboration:

message board calendar voting, …

Future plans: end systemsFuture plans: end systems

Embed in devices (game consoles, WinCE, appliances, …)

Wireless dedicated multimedia systems (wireless microphone, camera, …)

Languages for end-system service creation