Open source IP PBX

1. Introduction:

Open source IP PBX is based on the open source software (OSS) model. The software can be downloaded for free, and the source code is available to all users for modification and redistribution. Like other OSS licenses, General Public Licenses (GPLs) enable users to download and deploy the software without paying a license fee to the original developer. Unlike other OSS licenses, GPLs provide contributors with financial incentive that encourage active contribution to ongoing development.Cost savings, coupled with a rich and robust feature set and customization capabilities make open source IP PBX a viable option for your entire organization, especially for your contact center, which handles customer-facing interactions.

Converged TDM and packet-basedcommunications systems are a popular topic in computer science and telecommunications arenas. This research work discusses a system called Asterisk that is an open sourcehybrid TDM and packet voice platform.

Asterisk that is an open sourcehybrid TDM and packet voice platform. Asterisk isdesigned to interface any piece of telephony hardware orsoftware with any telephony application. This makesAsterisk a powerful component that can be easily used inNGN softswitches, conferencing servers and PrivateBranch eXchanges (PBX). Asterisk VoIP environment is integrated with H.323 and SIP packet-based networks.Its interworkingwith our H.323 and SIP networks and the functionality itoffers in terms of a converged TDM and packet-basedcommunications system.

Asterisk plays very important role to bridging the gap between traditional and networktelephony while Voice over IP (VoIP) is often thought of as little more than a method of obtainingfree long-distance calling, the real value ofVoIP is that it allows voice to become nothing more than another application in thedata network.

Asterisk's architecture is fundamentally simple, but differentfrom most telephony products. Essentially, Asterisk acts asmiddleware, connecting telephony technologies on thebottom to telephony applications on the top, creating aconsistent environment for deploying a mixed telephonyenvironment. Telephony technologies currently supportedby Asterisk include VoIP protocols like H.323, SIP, IAXand MGCP (both gateways and phones), as well as more

traditional TDM technologies like ISDN (PRI and BRI) andPSTN. Telephony applications include call bridging,conferencing, voicemail, auto attendant, custom interactive voice response (IVR) scripting, call parking, etc.

Asterisk is an open source software package. Hundreds, if not thousands, of developers are working every day on Asterisk, extensions of Asterisk, software for Asterisk, and customized installations of Asterisk. A big portion of the product's flexibility comes from the availability of the source code. This means, we can modify the behavior of Asterisk to meet our needs.

The challenge comes from the fact that an industry that has changed very little in thelast century shows little interest in starting now.

Research objective:

Integrating Asterisk (Open Source Telephony) as VOIP Gateway in Telecom Industry.

Research Questions:

* Why Asterisk is better than OtherVOIP Telephony Platforms?

* Why Traditional Telephony is switching towards the Voice over IP (VOIP)/data networks?

· What is the future of Open Source Telephony with Asterisk and its importance in VOIP Industry?

2. Literature Review:

I have collectedthis literature from electronic sources such as Google Scholar; different booksand web pages are also consulted with technology experts.

Background:

What is Voice over IP?

Voice over Internet Protocol (VoIP) is the basic method of carrying voice calls over an IPnetwork. This includes digitization and packetizationof voice streams. In other words, it is a mechanism of sending voice traffic over a data network.

Illustration below shows a basic (VoIP) setup.

Phone A can call Phone B as long as they are both connected to same data network. The basic idea behind VoIP is that it can send a telephone call over the same networks that carry data throughout a company, whether it be a local-area network (LAN), a corporate intranet, a wide-area network (WAN), or even the public Internet. To do so, the technology converts analog sound into tiny digital units called packets, then sends those packets over the network and reassembles them in the correct order on the receiving end .

What is IP Telephony?

IP Telephony uses VoIP standards to create telephony systems which offer all the features of a traditional private branch exchange (PBX) systems, as well as a range of integrated new software applications which can increase functionality and productivity.

These productivity applications, such as advance call routing, unified messaging, interactive voice response system, and call center applications are modular and tightly integrated through the use of industry standard protocols. Some examples of these protocols are SIP, H.323, MGCP, and SGCP. The definitions of these protocols are outside the scope of this study. To find out more about these you may visit IETF website.Illustration below is a basic IP Telephony setup:

Voice over Internet Protocol (VoIP) refers to voice communications that are delivered via an IP network, such as packet-switched networks or the Internet. The advent of this technology offers businesses myriad benefits, but many businesses are reluctant to completely overhaul their analog systems. The cost and labour associated with a full-VoIP replacement can be mitigated with the implementation of a Hybrid VoIP system, which offers the functionality of a full VoIP system without the extensive investment.

The Benefits of VoIP

Because VoIP converts an analog voice signal into digital format, telephone conversations essentially represent data transfers via Internet Protocol (IP) packets. These data packets can be sent with considerably less expense than traditional analog signals. Furthermore, the flexibility of data transmission means that VoIP technology offers a wide range of capabilities. Therefore VoIP provides several cost and usability benefits:

* Reduction in communication costs, especially for long-distance and international calls

* Simplified office wiring, with only one set of wires at each workstation, for both data and voice applications

* IP-based technology that allows sales and other personnel to share content, email, and files while conferencing

* Unified messaging, so that voicemail can be accessed from workstation handsets, soft phones, or PC

* Access to the Private Branch Exchange (PBX) from outside the office, for employees who travel or telecommute

* Seamless, update-free employee transfer among offices, since the phone number transfers with the IP phone set

VoIP offers a single dialing and voicemail system that allows free calls among office sites and supports remote workers. These appealing options make VoIP an attractive business communication solution. Yet people remain hesitant to switch from PBX to full VoIP, due to several factors:

* The prospect of replacing equipment that still functions and retraining employees presents considerable expense.

* If the existing LAN network does not have sufficient excess capacity, rewiring will be necessary, to ensure the quality of voice and data transmission.

* Disruption to business during these upgrades can result in lost revenue.

* It is somewhat difficult to quantify and measure the financial benefits of the productivity benefits and cost benefits of VoIP.

What is Asterisk?

Asterisk LogoAsterisk is the world's most popular open source telephony project. Under development since 1999, Asterisk is free, open source software that turns an ordinary computer into a feature-rich voice communications server. Asterisk makes it simple to create and deploy a wide range of telephony applications and services.

Code for Asterisk, originally written by Mark Spencer of Digium, Inc., has been contributed from open source software engineers around the world. Currently boasting over two million users, Asterisk supports a wide range of telephony protocols. It includes rich support for the handling and transmission of voice over traditional telephony interfaces including analog lines, ISDN-BRI lines and digital T1/E1 trunks. Asterisk also features support for a wide range of VoIP protocols including SIP, IAX and H.323 among others. It supports U.S. and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.

Asterisk is released as open source under the GNU General Public License (GPL), and it is available for download free of charge. Asterisk® is the leading open source telephony project and the Asterisk community has been ranked as a key factor in the growth of VoIP.

What Does Asterisk Do?

Asterisk is like an erector set or a box of Legos for people who want to create communications applications. That's why we refer to it as a "tool-kit" or "development platform". Asterisk includes all the building blocks needed to create a PBX system, an IVR system or virtually any other kind of communications solution. The "blocks" in the kit include:

* Drivers for various VoIP protocols.

* Drivers for PSTN interface cards and devices.

* Routing and call handling for incoming calls.

* Outbound call generation and routing.

* Media management functions (record, play, generate tone, etc.).

* Call detail recording for accounting and billing.

* Transcoding (conversion from one media format to another).

* Protocol conversion (conversion from one protocol to another).

* Database integration for accessing information on relational databases.

* Web services integration for accessing data using standard internet protocols.

* LDAP integration for accessing corporate directory systems.

* Single and mult-party call bridging.

* Call recording and monitoring functions.

* Integrated "Dialplan" scripting language for call processing.

* External call management in any programming or scripting language through Asterisk Gateway Interface (AGI)

* Event notification and CTI integration via the Asterisk Manager Interface (AMI).

* Speech synthesis (aka "text-to-speech") in various languages and dialects using third party engines.

* Speech recognition in various languages using third party recognition engines.

This combination of components allows an integrator or developer to quickly create voice-enabled applications. The open nature of Asterisk means that there is no fixed limit on what it can be made to do. Asterisk integrators have built everything from very small IP PBX systems to massive carrier media servers.

Asterisk Key Applications:

Asterisk as A PBX

Asterisk can be configured as the core of an IP or hybrid PBX, switching calls, managing routes, enabling features, and connecting callers with the outside world over IP, analog (POTS), and digital (T1/E1) connections.

Asterisk runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris and provides all of the features you would expect from a PBX including many advanced features that are often associated with high end (and high cost) proprietary PBXs. Asterisk's architecture is designed for maximum flexibility and supports Voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk asA Gateway

It can also be built out as the heart of a media gateway, bridging the legacy PSTN to the expanding world of IP telephony. Asterisk's modular architecture allows it to convert between a wide range of communications protocols and media codecs. Asterisk as a feature/media server.

Asterisk in The Call Center

Asterisk has been adopted by call centers around the world based on its flexibility. Call center and contact center developers have built complete ACD systems based on Asterisk. Asterisk has also added new life to existing call center solutions by adding remote IP agent capabilities, advanced skills-based routing, predictive and bulk dialing, and more.

Asterisk inThe Public Network

Internet Telephony Service Providers (ITSPs), competitive local exchange carriers (CLECS) and even first-tier incumbents have discovered the power of open source communications with Asterisk. Feature servers, hosted services clusters, voicemail systems, pre-paid calling solutions, all based on Asterisk have helped reduce costs and enabled flexibility.

Supported platforms

Asterisk® is primarily developed on GNU/Linux for x/86 and runs on GNU/Linux for PPC along with OpenBSD, FreeBSD, and Mac OS X. Other platforms and standards-based UNIX-like operating systems should be reasonably easy to port for anyone with the time and requisite skill to do so.

Asterisk® is available in Debian Stable and is maintained by the Debian VoIP Team

Supported hardware

Asterisk® needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk® supports a number of hardware devices, most notably all of the hardware manufactured by Digium®, the creator of Asterisk®.

Supported protocols

Asterisk® supports a wide range of protocols for the handling and transmission of voice over traditional telephony interfaces including H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP).

Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol Asterisk® merges voice and data traffic seamlessly across disparate networks. The use of Packet Voice allows Asterisk® to send data such as URL information and images in-line with voice traffic, allowing advanced integration of information.

Asterisk® provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.

AsteriskFeatures

Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk® offers both classical PBX functionality and advanced features which interoperates with traditional standards-based telephony systems and Voice over IP systems.

Call Features

ADSI On-Screen Menu System
Alarm Receiver
Append Message
Authentication
Automated Attendant
Blacklists
Blind Transfer
Call Detail Records
Call Forward on Busy
Call Forward on No Answer
Call Forward Variable
Call Monitoring
Call Parking
Call Queuing
Call Recording
Call Retrieval
Call Routing (DID & ANI)
Call Snooping
Call Transfer
Call Waiting
Caller ID
Caller ID Blocking
Caller ID on Call Waiting
Calling Cards
Conference Bridging
Database Store / Retrieve
Database Integration
Dial by Name
Direct Inward System Access
Distinctive Ring
Distributed Universal Number Discovery (DUNDi™)
Do Not Disturb
E911
ENUM
Fax Transmit and Receive (3rd

Party OSS Package)
Flexible Extension Logic
Interactive Directory Listing
Interactive Voice Response (IVR)
Local and Remote Call Agents
Macros
Music On Hold
Music On Transfer:
- Flexible Mp3-based System
- Random or Linear Play
- Volume Control

Call Features

Predictive Dialer
Privacy
Open Settlement Protocol (OSP)
Overhead Paging
Protocol Conversion
Remote Call Pickup
Remote Office Support
Roaming Extensions
Route by Caller ID
SMS Messaging
Spell / Say
Streaming Media Access
Supervised Transfer
Talk Detection
Text-to-Speech (via Festival)
Three-way Calling
Time and Date
Transcoding
Trunking
VoIP Gateways
Voicemail:
- Visual Indicator for Message Waiting
- Stutter Dialtone for Message Waiting
- Voicemail to email
- Voicemail Groups
- Web Voicemail Interface
Zapateller

Computer-Telephony Integration

AGI (Asterisk Gateway Interface)
Graphical Call Manager
Outbound Call Spooling
Predictive Dialer
TCP/IP Management Interface

Scalability

TDMoE (Time Division Multiplex over Ethernet)
Allows direct connection of Asterisk PBX
Zero latency
Uses commodity Ethernet hardware
Voice-over IP
Allows for integration of physically separate installations
Uses commonly deployed data connections
Allows a unified dialplan across multiple offices

Codecs

ADPCM
G.711 (A-Law & μ-Law)
G.722
G.723.1 (pass through)
G.726
G.729
GSM
iLBC
Linear
LPC-10
Speex

VoIP Protocols

SIP (Session Initiation Protocol)
IAX™ (Inter-Asterisk Exchange)
H.323
MGCP (Media Gateway Control Protocol
SCCP (Cisco® Skinny®)

Traditional Telephony Protocols

E&M
E&M Wink
Feature Group D
FXS
FXO
GR-303
Loopstart
Groundstart
Kewlstart
MF and DTMF support
Robbed-bit Signaling (RBS) Types
MFC-R2 (Not supported. However, a patch is available)

PRI Protocols

4ESS
BRI (ISDN4Linux)
DMS100
EuroISDN
Lucent 5E
National ISDN2
NFAS
Q.SIG

3. Research Methods:

The purpose of this chapter is to provide reader a brief introduction to the research approach and methods. A method is a tool, a way to solve a problem and research new knowledge. In this chapter, the procedure of the research will be presented.

In order to achieve the aim and objectives mentioned earlier there will be two methods used, primary data collection method and secondary data collection method. The primary method has been mentioned earlier as conducting interviews and survey from the different users to meet the requirement of Research objective. The secondary method will be to use a case study based approach and has been mentioned below with support from literature.

So for Asterisk and its impact on VOIP industry we have to conduct survey for data gathering from different Telecom Organization whether they are supplying Hardware or Software in VOIP Sector. Also gather data about those companies which are using Asterisk as telephony platform.

Many researchers have supported the use of the case study method and have suggested the steps one should consider in order to complete the literature review successfully. In order to achieve the aim and objectives, the following steps are important.

* Data Gathering (Interviews, Surveys etc)

* Analysis and study of Different Open Source VOIP Platforms.

* Validity and reliability (Evaluate information on basis of cost, technology and quality )

· Document findings

4. References:

Jim V, Leif M (2004). Asterisk™: The Future of Telephony. USA: O'Reilly.p 01-09

David G (2005). Building Telephony Systems with Asterisk. UK: Packt Publishing. p05-15

Penton, A. (2004). Asterisk: A Converged TDM and Packet-based. Voice over IP.01 (01), p01-05.

Anonymous . (2009). Asterisk (PBX). Available: http://en.wikipedia.org/wiki/Asterisk_(PBX). Last accessed 11-Dec-2009.

M. Spencer, M. Allison, C. Rhodes, The Asterisk Handbook, 2003,

Asterisk Documentation Team, Available online at www.asterisk.org.

Arsalan H. Abbasi. (2003). Voice over IP.A Discussion of Business and IT Challenges.1 (1), p04-06.

Kristin Masters. (2009). PBX to VoIP. Available: http://www.powersourceonline.com/magazine/2009/12/pbx-to-voip. Last accessed 15 JAN 2010.

Jody, V. (2009).Future Telephony Platforms. Available: http://www.powersourceonline.com/magazine/2009/01/future-telephony-platforms. Last accessed 18 JAN 2010.

David ,B (2009). Build feature-rich telephony systems with Asterisk. Birmingham: Packt Publishing. p7.

Please be aware that the free essay that you were just reading was not written by us. This essay, and all of the others available to view on the website, were provided to us by students in exchange for services that we offer. This relationship helps our students to get an even better deal while also contributing to the biggest free essay resource in the UK!