Report on VoIP
by admin on Mar.03, 2011, under Asterisk
Report on VoIP
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 network, 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.
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