IP PBX using SIP

Voice over Internet Protocol

Key Components for an IP PBX
setup
Wireless/Fiber IP Networks (Point to
point/multi point, LAN/WAN/Internet)
Central or Multicast SIP Proxy/Server
based Virtual IP PBX
IP Phone/USB Phone/Soft Phone as
origination/termination CPE (Client
Premise Equipment)
Basic Core Features & Advantages
of IP telephony
One time deployment cost, recurring billing mechanism completely
eliminated

Built in Features include automated IVR call attendance, Voice Mail,

Call Routing, Group Calling, Call Hold and many more

Detail CDR (Call Detail Logs) and built in Call Taping Management

Can be integrated to any PSTN, Analog, GSM networks using FXO

adapter, so that Definite calls can also be routed to normal &
commercial telephony exchange

Foreign UN Mission Team can have Voice Communication with the

HQ or any other point of presence
Understanding SIP Basics:

SIP is the Session Initiation Protocol- an ITU Protocol &

Standard
In the world of VoIP, SIP is a call setup protocol that
operates at the application layer
SIP can also be used to set up video and audio multicast
meetings, or instant messaging conferences
SIP is a major upgrade over protocols such as the Media
Gateway Control Protocol (MGCP), which converts
PTSN audio signals to IP data packets.
SIP History & Developments
SIP emerged in the mid-1990s from the research of Henning
Schulzrinne, Associate Professor of the Department of Computer
Science at Columbia University, and his research team.

As early as 2001, vendors began to launch SIP-based services.

To date, the 3G Community has selected SIP as the session control

mechanism for the next generation cellular network.

Microsoft has chosen SIP for its real-time communications strategy

and has deployed it in Microsoft XP, Pocket PC and MSN
Messenger.

Vonage, a service provider targeting consumer and small business

customers, delivers over 20,000 lines of digital local and long
distance calling and voice mail to over customers using SIP.
Access Modes to IP PBX System
SIP Proxy/Server Virtual IP PBX:

the SIP proxy only participates in the SIP

user authentication (Radius) and
messages---once the call is set up, the
phones send their voice traffic directly to
each other without involving the proxy
SIP proxies are very helpful in offloading
tasks and simplifying implementation of
end station telephones
Schematic Diagram
PRE CALL SCENERIO

- User Authentication Management

- Group management
- Call Routing or Re Routing
- Called Party Call Establishtment

IP Phone
IP Netowrk
IP Netowrk

IP Phone
Called Party
SIP Server providing IP
Calling Party PBX

POST CALL SCENERIO

IP Phone
IP Netowrk
Once the call is
established Proxy
IP Phone terminates, provided that
call taping mechanism not Called Party
activated
Calling Party SIP Server providing IP
PBX
 Function Description

User location End points (telephones) notify SIP proxies of their location; SIP
and determines which end points will participate in a call.
registration

User SIP is used by end points to determine whether they will answer
availability a call.

User SIP is used by end points to negotiate media capabilities, such as

capabilities agreeing on a mutually supported voice codec.

Session setup SIP tells the end point that its phone should be ringing; SIP is
used to agree on session attributes used by the calling and called
party.

Session SIP is used to transfer calls, terminate calls, and change call
management parameters in mid-session (such as adding a 3-way conference).
Features of IP PBX
Virtual PBX Server- provides access
platform using IP/Soft/USB Phone
Call Recording System
Call Attendant System
Call on Hold Player
Virtual PBX Server- provides access
platform using IP/Soft/USB Phone

The software works as a fully featured telephone switch

connecting to phone lines and extensions using state-of-the-art
VoIP technology
Offering all the normal features of a traditional PBX such as
allowing internal or external calls and more advanced call
queuing for call center applications the software routes all calls
within a premise or defined group or segments
Includes a call queue sequencer with voice prompting and on-
hold messages player
Works with Any Standard Soft Phone (Free client Software is
bundled), USB Phone or IP Phone
Connects directly to the Call Recording Platform o record calls if
required.
Detail CDR is stored in the DB for future usage
Call Recording System
This audio recording software can record 1 to 32
audio channels simultaneously with automated start
and stop if required.
CRS features digital signal processing to improve
voice intelligibility and automatic level control.
The recordings are automatically compressed for
archiving. Later they can be searched by date, time,
line or other data using the software directly or even
using just your web browser (if you enable web
access).
Answering Attendant Software (includes
Voice mail, call attendant, info line)
This software is an effective voicemail, call attendant,
info-line, audiotext or autodial solution
It can redirect in-coming calls during office hours or
act as a PC answer machine and take messages for
a number of voice mail boxes after hours. All calls
(including those answered by you) are logged with
date, time and caller ID. The recorded messages can
be played at any time, forwarded to an email address,
accessed via the internet or, if necessary, saved for
future reference.
Call on Hold Player

o This software mixes and plays messages and

music that will play to your callers while they
are on-hold or being transferred.
Client Premise Equipment (CPE)
IP Phone
Soft Phone

USB Phone
IP Phone
Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP,
DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP,
NTP
Support NAT traversal (STUN, etc), server fail-over, SIP presence
(SIMPLE), and more
ultiline support of up to 11 lines indicators (expandable to a few dozen more
through expansion key-module)
Graphical LCD to display up to 8 lines and 22 characters per line
Dual 10/100Mbps Ethernet ports
Headset jack
Support Caller ID display or block, per call or permanent
Call waiting, Hold, Mute, Transfer (blind or attended), Forward, and more
Multi-party conferencing
Integrated Power-over-Ether (802.3af)
And many more enterprise grade features
USB Phone
Commercial grade high quality speakerphone.
Large LCD display with backlight.
Selectable ring style and volume for incoming
calls.
Caller ID display.
Echo cancellation, noise reduction, full duplex
communication
PC-to-PC, PC-to-phone, Phone to Phone
operation
Soft Phone
Lets you make internet phone calls free direct PC to PC, or PC to phone via a VoIP
SIP gateway provider.
Supports up to 6 lines on the one phone with the ability to put calls on hold.
Works with a headset or in speakerphone mode with just a standard microphone and
set of speakers.
Includes data compression (GSM, uLaw, ALaw, PCM and G726), echo cancellation,
noise reduction, comfort noise and more.
Uses the standard SIP protocol so it can link to a broad range of telephone gateways,
SIP systems or other internet phone software.
Can be configured to work behind NATs and Firewalls.
Supports caller ID display and logging.
Includes a phone book with quick dial.
Supports call transfer
Lets you record phone calls to wav
Allows up to 6 people to join one call using the Call conferencing feature
Allows for quicker and easier communication using the Push to talk intercom
Includes Do not disturb button
IP PBX- Admin Console