VOIP Protocol Testing Tool
computer science topics|
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Joined: Jun 2010
15-06-2010, 04:53 PM
Traditional telephones use cables that are connected to telephone exchange .the calls are first transmitted to the nearest phone exchange which is then send to the telephone exchange to the area belonging to the person to whom we are calling. The whole process is charged by the service provider. Whereas VoIP (Voice Over Internet Protocol) enables voice calls to be carried over the Internet or other networks designed for data. The fact that VoIP transmits voice as digitized packets over the Internet means that it has the potential to converge with other digital technologies, which in turn will result in new services and applications becoming available. The wonderful thing about this is that it does not cost you anything beyond your normal ISP costs while the telephones cost much more because, telephonic system involves the establishment of base stations, maintenance of base station and more of that each call needs to be the allocation of a particular frequency.
The purpose is on developing IP-based voice communication solutions because we believe that it is the future of telecommunications. VoIP signaling protocols have been rapidly evolving and standardization continues to be elusive. As a consequence, multiple protocols and their countless variations are the reality.
The main aim of this project and implimentation is to meet these protocol testing challenges. The project and implimentation focuses on testing on the VoIP protocol. VoIP is the technology used to transmit voice conversations over a data network using the Internet Protocol. The data network can be the Internet, local network, or combination thereof such as a virtual private network (VPN). Basically, by creating a messenger window in which the detailed information is required for the communication and is tested at the two ends. It deals with transaction of messages, initially when the call is made, when the call is ended and also during the duration of the call. To set up the connection between the two end points we require the IP addresses and ports. The VC++ has been used for designing the front end of the project and implimentation. The Session Initiation protocol (SIP) is also used that enables the flow of voice messages. Initially the call initiates using SIP protocol. SIP is the connection that connects you to other end. Most of this is behind the scenes and requires no intervention on user part. SIP proxies are simple devices. Once a call is connected the end devices can exchange data directly. The proxy do not have to be on the path and they are only involved in the setting (and later the ending) of the session.
VoIP involves single network that replaces the current set-up of twin, separate networks of voice (PBX) and data (LAN). This provides some immediate benefits such as increased reliability and reduced total cost of ownership, as well as additional functionality such as portability in which incoming phone calls are automatically routed to the VoIP phone, regardless of where it is connected on the network.
VoIP is to offer features that improve on those offered by traditional PBX systems. VoIP offers some of the distinctive features that are not provided by traditional PBX system such as voice and e-mail, instant messaging and video. Moreover, it enables those with disabilities to access their message through voice, audio or a combination of both. Hearing-impaired people can place or receive calls from their computer without the need for a legacy TTY device.
The main benefit, apart from replacing expensive mobile phone calls with cost-effective Internet-based calls, is that with wireless VoIP the user, once logged into the network, can access their usual phone extension, with all its usual functions and features, regardless of their actual location.
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