IP PBX using SIP. Voice over Internet Protocol

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1 IP PBX using SIP Voice over Internet Protocol

2 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)

3 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

4 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.

5 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.

6 Access Modes to IP PBX System

7 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

8 Schematic Diagram PRE CALL SCENERIO - User Authentication Management - Group management - Call Routing or Re Routing - Called Party Call Establishtment IP Netowrk IP Netowrk IP Phone IP Phone Calling Party SIP Server providing IP PBX Called Party POST CALL SCENERIO IP Phone IP Phone Calling Party IP Netowrk SIP Server providing IP PBX Once the call is established Proxy terminates, provided that call taping mechanism not activated Called Party

9 Function User location and registration User availability Description End points (telephones) notify SIP proxies of their location; SIP determines which end points will participate in a call. SIP is used by end points to determine whether they will answer a call. User capabilities SIP is used by end points to negotiate media capabilities, such as agreeing on a mutually supported voice codec. Session setup 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. SIP is used to transfer calls, terminate calls, and change call parameters in mid-session (such as adding a 3-way conference).

10 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

11 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 onhold 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

12 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).

13 Answering Attendant Software (includes Voice mail, call attendant, info line) This software is an effective voic , 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 address, accessed via the internet or, if necessary, saved for future reference.

14 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.

15 Client Premise Equipment (CPE) IP Phone Soft Phone USB Phone

16 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

17 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

18 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

19 IP PBX- Admin Console

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