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ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipd

Proxy, Redirect, Registration server.• Authentication• Programmable (SIP- CGI)

OpenSource SQL database: MySQL

http://www.mysql.com

User information:• Contact location• Profile (e.g., password)• Aliases• Conferences• Address book

System information• Configuration

ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipd

Web based configuration

Web server

User profile can be modified using web browser. • Creating new user (admin/normal)• Changing profile and contact information (“follow me” service).• Web CGI scripts• Both sipd and web scripts use the database

ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipd

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

Web based configuration

Web server

IP phones connected to the departmental LAN. Users are identified by id, e.g., “hgs@cs.columbia.edu”

Software (sipc) for desktop. Allows audio, video, chat, white board, device control, instant message, presence and desktop sharing. Also plugnsip.

ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipd

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

Phones128.59.19.233

Web based configuration

Web server

Phones register themselves with sipd when powered up.

Sipd stores the contact information in the database table:hgs@cs.columbia.edu => hgs@128.59.19.233

There can be multiple contacts. All registered phones ring, and the first to pick up is connected.

Regsitration can also be altered from the web interface

ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipd

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

Phones128.59.19.233

Web based configuration

Web server

(2) Phone rings, the user picks up the call and can talk to the caller.

(1) When somebody calls hgs@cs.columbia.edu, sipd gets the INVITE message and forwards the call to the current location.

Another IP phone

Based on user profile, sipd may ask for caller authentication.

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipdT1/E1 RTP/SIP

Telephone

SIP/PSTN Gateway

Department PBX Web based

configuration

Web server

Cisco 2600 router with SIP/PSTN gateway connects the departmental LAN with the PBX.

Departmental PBX (Nortel Meridian) connects both internal and external lines to the gateway.

Telephoneswitch

Internal T1/CAS

External T1/CAS

(Extension:713x)

Dial “8” to reach outside line

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipdT1/E1 RTP/SIP

Telephone

SIP/PSTN Gateway

Department PBX Web based

configuration

Web server

(2) The gateway forwards all PSTN calls to sipd; sip:7132@128.59.19.141

(1) When PBX receives a call for 9397132, it forwards the call to extension 7132. 7130-7139 is assigned to the gateway.

Telephoneswitch

Internal T1/CAS

(Extension:713x)

Dials 9397132

128.59.19.141

(3) Sipd looks into the dialplan, finds a mapping 7132=>hgs@cs.columbia.edu and forwards the call to the current location of “hgs”.

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipdT1/E1 RTP/SIP

Telephone

SIP/PSTN Gateway128.59.19.28

Department PBX Web based

configuration

Web server(2) Sipd authenticates the caller

and checks permissions. Sipd maps the number 5551212 to 85551212@128.59.19.28, adding the prefix “8” and the gateway address.

(3) PBX forwards the call to external line.

Telephoneswitch

128.59.19.141

(1) The IP user dials “sip:5551212@cs.columbia.edu”.

Dial “8” to reach outside lineExternal T1/CAS

(4) PSTN user receives the call.

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipdT1/E1 RTP/SIP

Telephone

SIP/PSTN Gateway

Department PBX Web based

configuration

Web server

Can use no-DID (direct inward dialing) mode for more numbers.

Telephone number mappings and privileges modifiable from the web

Telephoneswitch

Internal T1/CAS

External T1/CAS

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipdT1/E1 RTP/SIP

Telephone

SIP/PSTN Gateway

Department PBX Web based

configuration

Web server

SNMP agent for SIP MIB. Allows remote monitoring and control of the SIP server. (e.g., prompt when an unauthorized registration is attempted)

Telephoneswitch

SNMP(Network Management)

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

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

Allows an H323 client (Netmeeting) to use the services of our SIP infrastructure.

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

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

Provides a voice mail and answering machine service to all the registered users. Has web interface for accessing voice mails.

rtspd

SIP/RTSPUnified

messaging

RTSP media server

sipum

Quicktime

RTSP clients

RTSPGeneric media server for playback and recording of messages. Can work with existing RTSP client, Apple’s QuickTime.

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

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

Centralized conferencing server for audio and video. Users can join from IP as well as PSTN.

SIP conference

server

sipconf

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

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

Netmeeting user dials “sip:meeting@cs.columbia.edu”

SIP conference

server

sipconf

SIP user dials “sip:meeting@cs.columbia.edu”

PSTN user dials 1-212-9397139

Sipd maps 7139=> meeting@128.59.19.196

128.59.19.196

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

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 gateway allows control of X10 devices. We can control a “lamp” using SIP DO message.

Device GW

X 10

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

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

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

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

SIP based Voice XML browser to allow access to voice mail and other services from touch tone phones

Performance measurement and scalability for very large call/session volumes (for sipd, sipconf and rtspd).

Accounting and billing services

Communication portal:Address book, calendar, message board, file sharing, conferencing, etc.

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

ArchitectureArchitecture

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

Single box

ArchitectureArchitecture

T1/E1 RTP/SIP

Telephone

External SIP/PSTN Gateway

Telephoneswitch

Single box

• All servers in a single box• Connects to a PSTN/SIP gateway• Users can use soft clients or hardware phones• Web based configuration and management

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