This document describes the architecture of an integrated voice over IP (VoIP) system. It includes a SIP proxy, redirect, and registration server that uses a SQL database for user profiles and authentication. It allows softphone and hardware phone clients to connect to the public switched telephone network via a SIP/PSTN gateway. A web server provides web-based configuration of users and services. The system allows calling between VoIP and PSTN users with number mappings and incorporates features like conferencing, voicemail, and device control.

1. Architecture
SIP proxy,
redirect
server
SQL
database
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

2. Architecture
SIP proxy,
redirect
server
SQL
database
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

3. Architecture
SIP proxy,
redirect
server
SQL
database
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.

4. Architecture
SIP proxy,
redirect
server
SQL
database
sipd
e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
Phones
128.59.19.233
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

5. Architecture
SIP proxy,
redirect
server
SQL
database
sipd
e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
Phones
128.59.19.233
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.

6. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/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.
Telephone
switch
Internal T1/CAS
External T1/CAS
(Extension:713x)
Dial “8” to reach outside line

7. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/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.
Telephone
switch
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”.

8. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/E1
RTP/SIP
Telephone
SIP/PSTN Gateway
128.59.19.28
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.
Telephone
switch
128.59.19.141
(1) The IP user dials
“sip:5551212@cs.columbia.edu”.
Dial “8” to reach outside line
External T1/CAS
(4) PSTN user
receives the call.

9. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/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
Telephone
switch
Internal T1/CAS
External T1/CAS

10. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/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)
Telephone
switch
SNMP
(Network
Management)

11. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/E1
RTP/SIP
Telephone
SIP/PSTN Gateway
Department
PBX Web based
configuration
Web server
Telephone
switch
SNMP
(Network
Management)
SIPH.323
convertor
NetMeeting
siph323
H.323
Allows an H323 client
(Netmeeting) to use the
services of our SIP
infrastructure.

12. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/E1
RTP/SIP
Telephone
SIP/PSTN Gateway
Department
PBX Web based
configuration
Web server
Telephone
switch
SNMP
(Network
Management)
SIPH.323
convertor
NetMeeting
siph323
H.323
Provides a voice mail and
answering machine service
to all the registered users.
Has web interface for
accessing voice mails.
rtspd
SIP/RTSP
Unified
messaging
RTSP media
server
sipum
Quicktime
RTSP clients
RTSP
Generic media server for
playback and recording of
messages. Can work with
existing RTSP client,
Apple’s QuickTime.

13. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/E1
RTP/SIP
Telephone
SIP/PSTN Gateway
Department
PBX Web based
configuration
Web server
Telephone
switch
SNMP
(Network
Management)
SIPH.323
convertor
NetMeeting
siph323
H.323
rtspd
SIP/RTSP
Unified
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

14. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/E1
RTP/SIP
Telephone
SIP/PSTN Gateway
Department
PBX Web based
configuration
Web server
Telephone
switch
SNMP
(Network
Management)
SIPH.323
convertor
NetMeeting
siph323
H.323
rtspd
SIP/RTSP
Unified
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

15. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/E1
RTP/SIP
Telephone
SIP/PSTN Gateway
Department
PBX Web based
configuration
Web server
Telephone
switch
SNMP
(Network
Management)
SIPH.323
convertor
NetMeeting
siph323
H.323
rtspd
SIP/RTSP
Unified
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

16. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/E1
RTP/SIP
Telephone
SIP/PSTN Gateway
Department
PBX Web based
configuration
Web server
Telephone
switch
SNMP
(Network
Management)
SIPH.323
convertor
NetMeeting
siph323
H.323
rtspd
SIP/RTSP
Unified
messaging
RTSP media
server
sipum
Quicktime
RTSP clients
RTSP
SIP
conference
server
sipconf
Device GW
X 10

17. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/E1
RTP/SIP
Telephone
SIP/PSTN Gateway
Department
PBX Web based
configuration
Web server
Telephone
switch
SNMP
(Network
Management)
SIPH.323
convertor
NetMeeting
siph323
H.323
rtspd
SIP/RTSP
Unified
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.

18. e*phone
sipc
Software SIP
user agents
Hardware
Internet (SIP)
phones
Architecture
SIP proxy,
redirect
server
SQL
database
sipd
T1/E1
RTP/SIP
Telephone
SIP/PSTN Gateway
Department
PBX Web based
configuration
Web server
Telephone
switch
SNMP
(Network
Management)
SIPH.323
convertor
NetMeeting
siph323
H.323
rtspd
SIP/RTSP
Unified
messaging
RTSP media
server
sipum
Quicktime
RTSP clients
RTSP
SIP
conference
server
sipconf
Device GW
X 10
Single
box

19. Architecture
T1/E1
RTP/SIP
Telephone
External SIP/PSTN Gateway
Telephone
switch
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