IP PBX using SIP
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
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
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
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
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
IP Netowrk SIP Server providing IP
PBX
- User Authentication Management - Group management - Call Routing or Re Routing - Called Party Call Establishtment
SIP Server providing IP PBX
Once the call is established Proxy terminates, provided that call taping mechanism not activated IP Phone IP Netowrk IP Netowrk IP Phone Calling Party Called Party IP Phone IP Phone Calling Party Called Party PRE CALL SCENERIO
POST CALL SCENERIO
SIP is used to transfer calls, terminate calls, and change call parameters in mid-session (such as adding a 3-way conference).
Session management
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 setup
SIP is used by end points to negotiate media capabilities, such as agreeing on a mutually supported voice codec.
User capabilities
SIP is used by end points to determine whether they will “answer” a call.
User availability
End points (telephones) notify SIP proxies of their location; SIP determines which end points will participate in a call.
User location and
registration
Features of IP PBX
Virtual PBX Server- provides access
platform using IP/Soft/USB Phone
Call Recording System
Call Attendant System
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.
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
Answering Attendant Software (includes
Voice mail, call attendant, info line)
This software is an effective voicemail, call attendant,
info-line, audiotext or autodial solution
Call on Hold Player
o
This software mixes and plays messages and
Client Premise Equipment (CPE)
IP Phone
Soft 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
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
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
