Linux Community Magazine

1.0 Introduction on VoIP:

VoIP is a method to build applications utilizing software and devices. It is just like that network, which carries it, and hence it is not an application. In a VoIP there can be special VoIP server hardware like the analog telephone adapter (ATA), or there can be programmable services that work like a PBX. According to this all the VoIP components must work for the protocol conversation that make the voice, human telephone conversations happen. This means all the VoIP components should follow the same protocol to make that happen.

VoIP is a protocol used to transmit voice over Internet and other packet-switched internet networks. Other names frequently encountered and have the same meaning like the VoIP are IP telephony and Internet telephony, other names like voice over broadband, broadband telephony and broadband phone are given when the network being used is a broadband internet network.

VoIP systems interface with the PSTN for allowing of transparent telecommunication worlwide. There are benefits of VoIP like reducing infrastructure and communication costs as by routing calls on the existing internet networks and avoiding setting up of duplicate networks. Telephony speech signals like digital audio are passed by VoIP systems which are typically cheap in data rates by means of the techniques like communication and data compression. They are packetized into small segments of tens of milliseconds speech and are encapsulated in a packet to transmit it over IP.

Since the first computer networks were developed VoIP has been a area of interest. In 1973 the voice was being transmitted using the internet. In the early 1980s this technology was provided to end users for sending voice conversations on the internet. Then in 1996 VocalTel Internet Phone software which was shrink-wrapped provided VoIP with some additional features like voicemail and caller ID. But this software did not offer gateway to PSTN, hence it was only possible to speak to a user who had VocalTel internet phone. Then the development of first soft switch began, these were developed to replace hardware telephone switches and to serve as gateway between different telephone networks.

2.0 Function of VoIP:

The facilities provided by VoIP would have become more difficult for implementation and can be expensive by using PSTN.

VoIP has the ability to transmit multiple telephone calls using the same internet connection. The services like conferencing, call forwarding, automatic redialling and caller-ID provided by telecom companies would be gained for zero or near zero costs. The calls will be more secure using standardized protocols. Internet connection is needed to connect to a VoIP provider, hence it is location independent. Using VoIP integration with other services that are available on the Internet, which include video conversation, data file or message exchange in parallel to the conversation, audio conferencing, managing address book. Advanced features like computer screen popup, call routing and IVR implementations are much easier and cheaper to integrate and implement.

3.0 Implementation of VoIP:

Transmission of analog signal using switched circuit is the basis of today's PSTN. In contrast to this VoIP networks transmit digitized voice using packet-based systems. As we can see VoIP is providing telephony services at compelling prices. [Syngress, Jason Sinclair, 2002].

Toll Bypass:

In contrast to the internet, PSTN provide voice services with heavy charges or tolls. Toll bypass is one such technique to avoid PSTN charges by using data networks, such like the Internet for carrying voice.

3.1 VoFR

It is a use of frame relay system to transmit IP packets having digitized voice data. Hence it is know as voice over frame relay (VoFR). [Sharma, D., 2002]

3.2 VoATM

It is a use of ATM network to transmit digitalized voice packets. Instead of carrying different length frames an ATM system carries small and fixed length frames known as cells. Hence the name Voice over Asynchronous Transfer Mode (VoATM).

4.0 Security:

VoIP is totally based on network protocols, and it needs to be evaluated from the same perspective when considering the security for a VoIP system [Jim Van Meggelen, Leif Madsen and Jared Smith, 2007]. But this is not to say as to traditional telecom security should be not kept in mind but here we need to give some attention towards the underlying network.

4.1 Basic network security

Single of the majority effective thing that can be complete is to safe access to the voice network. The employ of firewalls and VLANs are example of how this could be achieve. By evade, the voice network be made-up to be reachable only to persons or things that are in need.

4.2 Segregating voice and data traffic

If there is a requirement to have voice and data on the similar network, there may be a few values in maintaining them separately.

4.3 DMZ

Insertion your VoIP network in a DMZ can give an extra layer of security for your

LAN, while unmoving allows connectivity for applicable applications. In spite of whether you set up within a DMZ, any irregular traffic coming out of the network should be suspected.

4.4 Server hardening

Hardening your Asterisk server is serious. Not just are at hand are performances benefits to work on this, the removal of something not necessary will decrease the possibility that an broken vulnerability in the operating method can be utilized to gain admission and start an attack on additional parts of your network. Making Asterisk as non-root is a vital part of system hardening.

4.5 Encryption

Even while Asterisk do not yet fully maintain SRTP, it is still likely to encrypt VoIP transfer. For example, in between sites a VPN might be working.

4.6 Physical security

Physical security must not be unnoticed. All terminate equipments (like switches, routers, and the PBX) should be safe in a situation that can just be accessed by approved persons.

5.0 The Software for VoIP Application:

Asterisk is utilized as an open basis software PBX. At first it was developed in 1999 to meet a telecommunication requires. What meant to be an experiment had progressed into a worldwide phenomenon that changes the face of telecommunication knowledge now and the future. Today, Asterisk is the de facto criterion for voice switching and PBX function in open source space [Jim Van Meggelen, 2007]. Approximately there are 2 million Asterisk servers worldwide and rising. Asterisk gives you real-time connectivity on together PSTN and VoIP networks linking employees working from home to the office PBX over broadband connections. Never in the record of telecommunications has a system so well-matched to the requirements if industry been obtainable, at any price. Asterisk is a facilitate technology and, as by means of Linux, it will turn into increasingly rare to locate an enterprise with the intention of is not running some edition of Asterisk, in some ability, wherever in the netwo rk, solve a problem as just Asterisk can. [Jim Van Meggelen, 2007]

5.1 Benefits of Asterisk Application

Cheap call rates. Administer your telephone arrangement. Fast and easy growth. Situation Integration with business system or existing PBX and VOIP for major savings. No matter of proprietary system Guarded and lithe dial plan. Cheap cost contact to advanced phone Features

5.2 Benefits for Customer

Lower use cost free customer from sole vendor dependency. Provide freedom in system implementation. Eliminate upgrade-path costs. [Phil Lam, 2008] No per-seat license costs. Provides substitute avenues for bear/big fixes. Charge saving on calls.

Asterisk was produced and developed by Digium, Inc. It is a organization based in Huntsville, Alabama who specializes in the development of PBX hardware and open-source telephony software, most notably Asterisk [Digium Inc, 2008]. Digium was founded by Mark Spencer as Linux Support Services in 1999. Contract support and development for Linux were provided by the organization. As Mark did not have the resource to purchase a PBX for his own company, he decided to start writing one from scratch. In 2001, the U.S. economy was going through a period of recession, and as a result demands for the services provided by the company dried up. Mark then took the step to shift the company's focus on its fledgling open-source PBX product, Asterisk. The company was given a new name "Digium" in 2002.

6.0 Asterisk's Requirements:

The source code of asterisk can be downloaded as it is accessable like any other open source softwares, and compile the code by yourself. Asterisk get compiled easily on Linux but it will not even run on FreeBDS, Solaris and Mac OS X. [Ted Wallingford, 2005]. Used in combination with Digul's telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures. Digium's offerings include VoIP, conferencing, voicemail, legacy PBX, IVR, auto attendant, media servers and gateways, and application servers and gateways.

Many engineers had contributed for the code of asterisk programmed by mark spencer of digium. The engineers were from around the globe. Presently boasting over two million users, wide range of TDM protocols are supported by Asterisk for the handling and transmission of voice over traditional telephony interfaces, featuring VoIP packet protocols such as SIP and IAX. Signaling types that are used in business phones are suppported by U.S and European standard, and they act as a bridge between new-generation voice-data integrated networks and present infrastructure.

7.0 SIP:

SIP [J. Rosenberg, H. Schulzrinne, Camarillo, Johnston, Peterson, Sparks, Handley, and Schooler, 2002.] is a direct (or signalling) protocol similar to HTTP. It is a protocol to can set up and tear along any type of conference. SIP call control uses SDP [M. Handley and V. Jacobson, 1998] to explain the particulars of the call. SIP uses a URI7 to recognize a logical destination, not an IP address. The address might be a nickname, an e-mail address, or even a telephone number. SIP can as well be utilized to send immediate text mails. In the 1960s and 1970s, dumb terminals are used to admission applications on a supercomputer shared by a lot of hundreds of clients. Opening in the 1980s, it began to employ refined application on a PC, but it was also able to employ the PC as a infrastructure terminal to gain admission to applications and database on joint computers (servers) in the network. SIP hosts with a variety of degrees of complexity spirit perform a little function loca lly while allow us to contact application in the network. SIP user can interact openly with these applications.

By using the client-server model, logical entities are defined by SIP that is being implemented jointly or separately in a product. Users send SIP needs, whereas servers admit SIP requests, carry out the request methods, and act in response. The SIP design defines six request methods:

REGISTER allows either the user or a third party to register contact information with a SIP server.

INVITE initiates the call signalling series.

ACK and CANCEL hold session set-up.

BYE terminates a session.

OPTIONS query a server concerning its capabilities.

Registrar is a SIP server that receives, authenticates and accepts REGISTER requests from SIP clients. It may be collocated with a SIP proxy server.

7.1 SIP Soft phones Used:

X-lite, X-Pro and Eyebeam:

Xten Networks invented several VoIP softwares and soft phones. No VoIP test is carried in lab is completed without an X-lite, the company SIP soft phone for windows, Mac, and thin-clients setups. Xten also invented X-Pro, a fully featured business SIP soft phone for Windows, Mac, and MS PocketPC. Xten also produces Eyebeam, a SIP video conference phone. [www.xten.com]

8.0 IAX:

The Inter-Asterisk Exchange Protocol (IAX), currently is in its second revision, is used as a signaling protocol for VoIP networks, just like SIP. End-point and trunk signaling are also provided by this protocol. The main difference in between IAX and the SIP signaling family is that IAX does not implement RTP as the packet mechanism. Instead, IAX has its own way of packaging encoded voice. IAX is also NAT-proof, so dozens or hundreds of simultaneous calls from behind a masquerading firewall will function correctly, just as HTTP does. IAX is implemented in a far-simpler and less application-exhaustive manner than SIP and H.323. It is really intended just for telephony applications, while H.323 and especiallly SIP, include far more extensibility. Therefore IAX is much more packed. All the implementations has been done with as little as 64kb of code. When any IAX client wants to register with the IAX server or proxy, UDP port will be utilized. When again a call is placed, th e same UDP port is used. When again a voice transmits this port is used again. The IAX attaches headers and metadata in each packet that will define what the packets purpose is and whether it has a payload attached to it or not. By this IAX distinguishes between signalling, registration, and voice packets. IAX is an independent protocol created by Mark Spencer rather than a recommended standard like SIP. The specification of IAX has been accepted by the VoIP community though it is proprietary in nature. As a result it is quiet well used in Digium's products. Asterisk the open source PBX implements IAX fully. An ATA is manufactures by Digium too which is IAX based. Digium is also working on manufacturing hadrphones.

8.1 IAX Softphones Used:

8.1.1 Firefly:

Firefly is a softphone produced by Virbiage that implements IAX signalling. It makes it a fine choice for asterisk-based systems or for utilizing services like IAXTel which support the IAX protocol. [www.virbiage.com/firefly]

8.1.2 IAXComm:

IAXComm is an open source softphone for Windows, Linux, and Mac OS X. It is supported by IAX only. [iaxclient.sourceforge.net/iaxcomm]

9.0 Peoples involved in Asterisk:

Mark Spencer / Founder and CTO

While Mark Spencer was a computer engineering student in Auburn University set up a company known as Digium in 1999 with linux support systems. As he was a student it was difficult for him to own PBX, hence with his knowledge of C coding and Linux PC he developed his own PBX for his company. This phenomenon was the starting of an open source known as Asterisk. [Digium Inc, 2008]

As Asterisk attained popularity, Spencer turned his company focus from Linux to supporting Asterisk and expanding up the telecom market. Today digium is the leading open source telecom provider in the world. It happened with Mark using the fledgling start and renaming his company. Mark is know or regarded as pioneer for open source telephony and he gives regular keynote addresses to huge audiences.

David Deaton / Vice President, Engineering

Company's vice president of engineering post was filled by David Deaton in March 2007. Deaton is accountable for overseeing the growth of new Digium products; build upon the achievement of Asterisk NOW and Asterisk Business Edition.

Digium, creator and main developer of Asterisk, the lead open source telephony platform, and Skype, the lead global Internet communications company, has announced the beta version of Skype for Asterisk, which will permit the integration of Skype functionality into Digium's Asterisk software and allow customers to make, receive and transfer Skype calls from within their Asterisk phone systems.

"Throughout our individual histories, Skype and Asterisk have each disrupted conventional communication methods through innovative, cost-effective solutions," said Stefan berg, vice president and general manager for Skype Telecom and Skype for Business. [Digium Inc, 2008]

Particularly, the beta version of Skype for Asterisk is an add-on channel driver module that integrates Skype Internet calling with Asterisk-based telephony products. Skype for Asterisk too complement small and mid-sized business users' existing services on conditional low rates for calling landline and mobile phones around the globe.

10.0 The benefit of Asterisk

10.1 Functionality:

Flexible features are offered by Asterisk based telephony. PBX functionality along with advanced features and interoperatability with standard-based telephony systems , voice over IP systems are offered by Asterisk. The features which large proprietary PBX offers like voicemail, auto attenedent, conference bridging, call queing and many more are done by Asterisk. [Net hawk Pvt. Ltd, 2008]

10.2 Scalability:

VoIP balance or grows much further easily than customary telephone/PBX systems. Adding additional VoIP phone only require connection like receive calls via the office VoIP PBX as condition they were in the office. Asterisk can easily bond two or more branch offices using the Internet. No puzzling codes to dial; all users have their own sole extension. You can just dial directly. As your company expands and you demand more capital from your VoIP systems, there is no requirement to buy new Asterisk software. Server hardware can be improved and/or replaced whereas still retaining your similar Asterisk software, configuration, extensions and voicemail.

10.3 Maintenance:

Routine maintenance and moves-adds-changes will be performed by remote ways as Asterisk is IP based by any friendly representative. Simply a phone can be unpluged from an old location and pluged into new location . graphical user interface makes it easy for adding new users or extensions. [Net hawk Pvt. Ltd., 2008]

10.4 Cost:

Net-Telligence Group has included the Asterisk VoIP platform into a reasonably priced, cost effective explanation that is feature rich, easy to use, constant, scalable and trustworthy.

11.0 REFERENCES:

Configuring Cisco Voice Over IP, Syngress, Jason Sinclair, Martin Walshaw, Paul Fong, Eric Knipp, David Grey, Michael E Flannagan, Published by Syngress, 2002

Digium Inc, 2008. Available at www.digium.com [Accessed 9 January 2009]

iaxclient.sourceforge.net/iaxcomm. Accessed 13 January 2009.

Jim Van Meggelen, Leif Madsen and Jared Smith, Asterisk: The Future of Telephony. 2nd Edition, Published by O'Reilly, 2007

Net hawk Pvt. Ltd., 2008. Available at http://nethawk.com.pk [Accessed 10 January 2009]

Phil Lam, 4 October 2008. Available at http://communications.angsulong.com/why-asterisk [Accessed 10 January 2009]

SDP: Session description protocol, M. Handley and V. Jacobson, IETF RFC 2327, 1998

SIP: Session initiation protocol v.2.0, J. Rosenberg, H. Schulzrinne, Camarillo, Johnston, Peterson, Sparks, Handley, and Schooler, IETF RFC 3261, 2002

Ted Wallingford, 2005. Switching to VoIP. Published by O'Reilly, 2005.

Voice Over Internet Protocol (VoIP), Bur Goode, Senior Member, IEEE, PROCEEDINGS OF THE IEEE, VOL. 90, NO. 9, SEPTEMBER 2002

VoP (voice over packet), Sharma, D.Potentials, IEEE Volume 21, Issue 4, Oct/Nov 2002

www.cisco.com. Accessed 11 January 2009.

www.virbiage.com/firefly. Accessed 10 January 2009.

www.xten.com. Accessed 12 January 2009.

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