©2000, Columbia University©2000, Columbia University

““A flexible architecture to support wide range of A flexible architecture to support wide range of multimedia communication applications, both clients and multimedia communication applications, both clients and servers”servers”

Presented by: Kundan SinghJoint work with Wenyu Jiang, Jonathan Lennox, Sankaran

Narayanan, Henning Schulzrinne and Xiaotao Wuat Columbia University

Physical layer

Link layer

Network (IPv4, IPv6)

Transport (TCP, UDP)

Application layer

H.323 RTSP RSVP RTCP

RTP

MediaG.711MPEG

SIP

SignalingQuality of service

Media transport

InternetTelephony

InternetRadio/TV

Messagingand Presence

Interactivevoice response

Unified messagingVideo

conferencing

Multimedia Communication Multimedia Communication ProtocolsProtocols

CINEMA modulesCINEMA modules

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

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

Our IP telephony test-bedOur IP telephony test-bed

SIP proxy,redirectserver

SQLdatabase

sipdT1/E1 RTP/SIP

Telephone

SIP/PSTN Gateway

Department PBX Web based

configuration

Web server

Telephoneswitch

SNMP(Network Management)

SIPH.323convertor

NetMeetingsiph323

H.323

rtspd

SIP/RTSPUnified

messaging

RTSP media server

sipum

Quicktime

RTSP clients

RTSP

SIP conference

server

sipconf

Device GW

X 10

W. Jiang, J. Lennox, H. Schulzrinne and K. Singh, “Towards Junking the PBX: Deploying IP Telephony". NOSSDAV 2001,

PSTN to IP CallPSTN to IP Call

PBXPSTN

External T1/CAS

Regular phone(internal)

Call 93971341

SIP server

sipd

Ethernet

3

SQLdatabase

4 7134 => bob

sipc

5

Bob’s phone

• Direct Inward Dial (DID) - direct and simple• No-DID - dial extension, supports more users

GatewayInternal T1/CAS(Ext:7130-7139)

Call 71342

713x is called a part of Coordinated Dial Plan (CDP) in a Nortel PBX

IP to PSTN CallIP to PSTN Call

Gateway(10.0.2.3)

3

SQLdatabase

2Use sip:85551212@10.0.2.3

Ethernet

SIP server

sipdsipc

1Bob calls 5551212

PSTN

External T1/CASCall 55512125

5551212

PBX

Internal T1/CASCall 85551212 4

Regular phone(internal, 7054)

Note: In this direction there is no distinction between DID and non-DID calls.

Layered Libraries

Transport layer (TCP/UDP)

RTPInterface

HTTP Message Parsing

RTSP transaction

SIP transactionClient Branch

RTSP API

RTSP server

SIPUA API

SIP proxy

Other Applications

User InteractionUser Interaction

• Web interface– Administration– User configuration

• Unified Messaging

– Notify by email

– rtsp or http

• Portal Mode– 3rd party IpTelSP

http://www.cs.columbia.edu/~kns10/research/cinemahttp://www.cs.columbia.edu/~kns10/research/cinema

• Inter-working between SIP and H.323 version 2.0• H.323 fast-start as well as normal call• Multiple simultaneous independent calls• Transparent media traffic• Unix as well as Windows• Built-in gatekeeper• Different dialing modes

http://www.cs.columbia.edu/~kns10/software/sip323http://www.cs.columbia.edu/~kns10/software/sip323

SIPSIP H.323H.323

Gatekeepersipc

K. Singh, H.Schulzrinne, "Interworking Between SIP/SDP and H.323". Proceedings of the 1st IP-Telephony Workshop (IPTel'2000), April 2000.

sipconfsipconf

sipcsipc

http://www.cs.columbia.edu/~kns10/software/sipconfhttp://www.cs.columbia.edu/~kns10/software/sipconf

SIP323SIP323

SIP/PSTNSIP/PSTN

• SIP based conferencing server• SIP/SDP and RTP/RTCP• Audio mixing• Play-out delay algorithm• Web based conference setup• G.711 A and Mu law, G.721,

DVI ADPCM• Multiple simultaneous

conferences

K. Singh, G.Nair and H.Schulzrinne, “Centralized Conferencing using SIP". Proceedings of the 2st IP-Telephony Workshop (IPTel'2001), April 2001.

SIP/RTSP based SIP/RTSP based unified messagingunified messagingvoice mail, answering machine, web based setup, email and web integration . . .

http://www.cs.columbia.edu/~kns10/software/sipumhttp://www.cs.columbia.edu/~kns10/software/sipum

Kundan Singh and Henning Schulzrinne, "Unified Messaging using SIP and RTSP". IP Telecom Services Workshop 2000, Sept 2000. Atlanta, Georgia.

SIP based voicemailSIP based voicemailWide range of applicability

Campus/corporate networkCampus/corporate network

sipumsipum

rtspd

Within a domain

InternetInternet

sipumsipum

rtspdExternal application service provider

VoiceXML is a language for specifying voice dialogs for interactive voice response systems. It is specified in XML.

SipVxmlSipVxml

PSTN

SIP user agentSIP user agent

SIP/PSTN gatewaySIP/PSTN gateway

Web serverWeb serverCGI, servlet, JSP

SIP based VoiceXML browser

SIP phoneSIP phone

            Media serverMedia server

Call Request

Fetch VoiceXML pages

Get streaming media

Press 1 to listen to next message, 2 to forward …

Performance measurement Performance measurement and Scalabilityand Scalability

• Busy hour call arrival (BHCA)• Requests per second (proxy)• Request turn-around time (proxy)• Participants per conference (sipconf)• Simultaneous media streams (rtspd)• DNS based scalability with server farms• Stateless proxy• Hierarchical conference servers• Redirect feature

http://www.sipstone.orghttp://www.sipstone.org

Services and Services and applicationsapplications

Multiparty ConferencingMultiparty Conferencing

Unified messaging,Unified messaging,voice mail and voice mail and answering machineanswering machine

SIP/VoiceXML browserSIP/VoiceXML browser(In progress)(In progress)

Real-time Media StreamingReal-time Media StreamingSIP/H.323 translationSIP/H.323 translation

Hardware SIP phonesHardware SIP phones

Instant messagingInstant messagingand presenceand presence(In progress)(In progress)

SIP-PSTN gatewaySIP-PSTN gateway(In progress)(In progress)

Software SIP clientsSoftware SIP clients

Development Libraries Development Libraries (User agent API, SIP Stack)(User agent API, SIP Stack)

Programmable SIP Programmable SIP servers (CGI, CPL)servers (CGI, CPL)

… moving from IP telephony to a real-time multimedia collaboration portal…