serubi: Deploying an IP PBX

Wednesday, November 2, 2016

Deploying an IP PBX

Executive Summary

The task of commissioning an IP PBX system at Janani publications which is an SME company with a staff of 25 employees. They have an existing CDMA system and they want to upgrade their communication system to increase productivity and reduce outgoing call costs.

Due to increased number of telephone calls originating within Janani publications, I recommend migrating to a full manageable and cost effective IP PBX.

My plan is to implement, 12 internal extensions with 4 outside PSTN lines. IP PBX that uses freely available open source software is to be utilized. The work would be completed within 16 days, and a period of 7 days for testing and training staff. The system will be online on successful completion of the above work. This project report is a detailed report on how this IP PBX will be implemented and commissioning at Janani publications.

The Cost of operating the existing CDMA system is measured using the current telephone bills and number of calls originated. The outcome shows that the high telephone costs and low operational output requires drastic improvements to the network and telecommunication system.

Introducing IPBX system will curtail the call costs while increase the call productivity by giving more call time for less cost allowing Janani publications to utilize the IPBX system to improve their business and marketing performance using the telephone system. A detailed project analysis and a budget forecast have been formulated in accordance with the project objectives to obtain a realistic project outcome.

Introduction

We chose an IP PBX in place of a conventional PBX due to its Low initial investment by utilization of freely available open source software, saving of outgoing call charges now possible via dialing a local IP extension, easy to use interface unlike conventional PBX’s, On the fly allocation of new users and managing of users, host of features previously not possible with a conventional PBX such as auto attendant for all calls, Conferencing, Efficient utilization of available resources.

An aforementioned telecommunication solution serves well for a company like Janani Publications due to the simplicity and cost effectiveness of this solution.

The proposed project is to deploy an IP PBX at Janani publications. Currently the establishment operates on individual CDMA phone connections for each section. Another intercom system is in service to contact each section from the manager’s office. By implementing the proposed IP PBX the establishment will benefit from the reduced time it takes to contact each section, the ability to contact an outside line from each workstation and the biggest advantage would be the cost of the whole project which is considerably less when compared to deploying a conventional PBX with 12 internal lines and 4 outside lines. Adding to those are other features that comes with IP PBXs such as call conferencing, auto attendant, etc.

Implementation process will start off with a complete assessment of the existing equipment, required hardware and software, estimated cost and location of equipment.

The proposed project is to deploy an IP PBX at Janani publications. Currently the establishment operates on individual CDMA phone connections for each section. Another intercom system is in service to contact each section from the manager’s office. By implementing the proposed IP PBX the establishment will benefit from the reduced time it takes to contact each section, the ability to contact an outside line from each workstation and the biggest advantage would be the cost of the whole project which is considerably less when compared to deploying a conventional PBX with 12 internal lines and 4 outside lines.. Adding to those other heap of other features that comes with IP PBX’s such as call conferencing, auto attendant, etc.

Implementation process will start off with a complete assessment of the existing equipment, required hardware and software, estimated cost and location of equipment.

The system will comprise of a free open source software based trixbox PBX server and we will be utilizing the existing analog phones with the use of adapters known as analog telephone adaptors (ATA). The ATA’s will register their presence and connect with the Trixbox PBX server to make and receive calls.

Background information

Janani publications represent an establishment where they undertake in publishing of weekly and monthly magazines consisting of various entertainment matters. In addition to that they also carry out all stages of publishing ranging from color separation work, Offset printing, plate making and typesetting work as per customer’s requirements. Company comprises of 7 sections which are the chairmen’s office, manager’s office, color separation section, graphic and typesetting section, Offset printing section, Packaging and preparation and plate making section.

The company as stated in the introduction own a standalone intercom system and individual CDMA phones for each section.
To contact another section a user currently has to make an outgoing call through their CDMA phone to another sections CDMA phone. This incurs costly peak hour call costs for each and every call. The intercom serves mainly as a one way communication system for manager.
As for the company’s internet requirements, it is handled by a 1 Mbps ADSL lines provided by Sri Lanka Telecom.

Aims Of the project

Aim of the project is to create a complete intercompany communication solution for Janani publications. By implementing the IP PBX we wish to create a quick and easy way to communicate among the sections creating a more efficient way when collaborating day to day work tasks. Plus the system will avail the users a whole set of features including
The current CDMA system incurs huge amounts of call costs each month. By implementing this project the call costs will come down to 0% since all calls originates within the company.
Reducing the time taken to initiate a call and to connect
The current phone systems takes a considerable amount of time initiating a call and to connect when needed to another section/person. The Proposed project will reduce call connect times by 90% since all calls will connect within the company.
Eliminating the intercom system and the installation of IP phone extensions to each and every section of the company
The trixbox server will have its own standalone Interactive Voice Response system so that the incoming callers will be able to leave a message to any member of the staff or any section
The servers Interactive Voice Response system will make it possible for the incoming callers to contact any section or any member of staff by just following the instructions played through the system
Introducing a cost effective way to make use of value added services such as Call conferencing, Call forwarding and Interactive Voice response System for incoming Calls.
trixbox server features wide range of call features including call conferencing which makes it possible for members of staff to keep quick meeting.
Making use of cheap VOIP service providers, members of staff are able to call international telephone numbers with 1/10 of the cost of an IDD call.
Flexible and easy scalability, adding or removing users with just a click.
Class 5 features such as Caller ID, Call Forwarding, Call Transfer, Speed Dial, Three Way Calling and much more.
All group services available including Call Center, Auto-Attendant, Attendant Console, Hunt Groups, Conferencing, and more.
System upgrades at no cost and portability for your existing numbers.
Connects each section within the IP PBX without the cost of expensive PSTN.
Secure System, phone service is restored immediately in a disaster recovery event.
Savings reduce capital investment and ongoing expenses with a low upfront implementation cost.

Methodology

The project is based on the concept of voice over IP protocol. The VOIP protocol allows digitized voice data to be carried over internet protocol. This in relation with management methodology such as the proposed IP PBX trixbox, makes it possible to utilize low cost personal computers along with free open source software based IP PBX servers to make such a cost effective communication system.

The IP PBX trixbox is based on SIP (Session initiated protocol) and it facilitates the IP PBX to initiate sessions with connecting extension phone or Analog telephone adaptors.

The network will comprise of a free open source software based trixbox PBX server and we will be utilizing the existing analog phones with the use of adapters known as analog telephone adaptors (ATA). The ATA’s will register their presence and connect with the Trixbox PBX server to make and receive calls.

Relevant theory

VoIP

Voice over IP (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks. Other terms frequently encountered and synonymous with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.

Internet telephony refers to communications services — voice, facsimile, and/or voice-messaging applications — that are transported via the Internet, rather than the public switched telephone network (PSTN). The basic steps involved in originating an Internet telephone call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is reversed at the receiving end.

VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. Codec use is varied between different implementations of VoIP (and often a range of codecs are used); some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codecs.

Voice over IP has been implemented in various ways using both proprietary and open protocols and standards. Examples of technologies used to implement Voice over IP include:

H.323
IP Multimedia Subsystem (IMS)
Media Gateway Control Protocol (MGCP)
Session Initiation Protocol (SIP)
Real-time Transport Protocol (RTP)
The Session Initiation Protocol has gained widespread VoIP market penetration, while H.323 deployments are increasingly limited to carrying existing long-haul network traffic.[citation needed]
A notable proprietary implementation is the Skype protocol.

For this project the open source server that we have chosen runs entirely using SIP. So detailed explanation on SIP follows

SIP

The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet

Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, and adding or deleting media streams. Other feasible application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer and online games.

SIP was originally designed by Henning Schulzrinne and Mark Handley starting in 1996. The latest version of the specification is RFC 3261 from the IETF Network Working Group.[1] In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular systems.

The SIP protocol is an Application Layer protocol designed to be independent of the underlying transport layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP).[2] It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).[3]

Figure 1 Example of SIP protocol
Go to link Download