Term Paper on "Voice Over Internet Protocol"

Term Paper 4 pages (1593 words) Sources: 1+

[EXCERPT] . . . .

Voice over Internet Protocol -VoIP is a type of hardware and software that helps individuals to perform telephonic conservations through the Internet. Voice signals gets converted into packets of data which are then send on shared, public lines thereby bypassing the fees to be paid to the conventional public switched telephone network -PSTN. (VoIP (Voice over Internet Protocol: 2004)

Advantages of VoIP:

Unlike in case of PSTN line, using VoIP voice conversations can be made absolutely free of cost. Apart from that, conversations can be performed with multiple persons also. An important advantage of VoIP and Net telephony is that is does away with the usage charges billed by the ordinary telephone service companies. Concurrently one is able to exchange data with persons while talking, sending images, graphs and videos. (VoIP Voice over Internet Protocol Communication)

Functioning:

In the first step, the voice of the caller is digitized, since in VoIP all transmissions must be digital. This digitization can be performed either by the telephone company, or by an Internet Service Provider -ISP, or by a Personal Computer. In the second step, using intricate algorithms, the digital voice is compressed and then divided into packets; and utilizing the Internet protocol, the packets are addressed and transmitted through the network to be reassembled in the right order at the destination point. Once again this task of reassembling can be carried out by a carrier and ISP or by the PC. In the third step, while in transit in the Internet, the data packets might be lost or got delayed, or errors might cause damage to the packets. Conventional methods of error correction will request retransmission of unusable or packets which have been lost, however, in case the sound which is sent is a real-time voice communication, in that situation that technique will not be functional, hence sophisticated error detection and corrective systems are employed to create sound to fill in the dropouts in sounds. (Voice-over-IP: The Future of Communications)

In the fourth step, after the packets are sent and appear at the destination point, the transmission is assembled and decompressed to restore the data nearly to the original form. According to the suggestion of this explanation, the technology that works seamlessly in case of sending data might be less than ideal for voice transmissions. The technology is advancing even though the quality of voice transmission by the use of packet technology is less advanced compared to a circuit switched connection, and hence the disparity in quality would usually be felt by any listener. With the advancement of IP technology, the quality edge for voice communications benefits by the circuit-switch will go down, but the majority of specialists view uniformity in quality as still a long possibility. (Voice-over-IP: The Future of Communications)

Regulatory Issues:

The technologies that employ the Internet and Internet Protocol -IP networks to transmit voice communications possess the capability to lessen the costs, favor innovation, and enhance access to communications services in the developing nations throughout the world. But for the regulators these new technologies pose challenges as they are not applicable within the regulatory model of the recent past which in a lot of nations has dealt voice and data services in different footing. Particularly, in end to end applications, the launching of VoIP there are questions regarding: collection of global service funds and responsibility to give global service; payment of access or other fees for settlement among local and long distance carriers and global carriers; quality of service; and the impact on the revenue of monopolies which continue to exist for long distance and international voice service. In spite of that there is a rising recognition in national and international bodies that these matters are required to be solved in a manner that permit IP technologies to be employed in case of voice applications. Lifting the regulatory barriers to the use of IP technologies in case of voice will be able to (i) progress the target of service that is affordable (ii) support the installation of broadband network and services that represent the future of communications in the period of digital convergence. (Voice-over-IP: The Future of Communications)

Session Initiation Protocol (SIP):

SIP is the Internet Engineering Task Force -IETF's norm for conferencing through multimedia over the Internet Protocol. SIP is ASCII based application layer control protocol -defined in RFC 2543, which can be employed to set up, maintain, and end calls between two or more terminal points. Similar to other VoIP protocols, SIP is devised to tackle the work of signaling and session management within a packet telephony network. Signaling permits call information to be carried through network boundaries. Session management gives the ability to manage the features of an end-to-end call. (Overview of the Session Initiation Protocol)

SIP provides the potential to: (i) find out the location of target end point -SIP favours address resolution, mapping of names and call redirection. (ii) Finding out the media capabilities of the target end point through Session Description Protocol -SDP, SIP finds out the lowest level of general services between the end points. Conferences are set up using only the media potential which can be supported by all end points. (iii) Determine the accessibility of target end points. In case a call cannot be finished due to the unavailability of target end point, SIP finds out whether the called party is already on the telephone and did not reply in the allotted number of rings. Subsequently, it returns a message stating why the target end point was not available. (iv) Set up a session between the starting and target end points. In case the call can be completed, SIP sets up a session between the terminal points. SIP also supports changes in the middle of the call like the addition of another end point to the conference or the changing of a media feature or codec. (v) Tackle the transfer and ending of the calls-through SIP transfer of calls from one end point to another can be done. At the time of call transfer, SIP just establishes a session between the transferee and a new end point and ends the session between the transferee and the transferring party. During the end of the call, SIP ends the session between all the parties. (Overview of the Session Initiation Protocol)

Components of SIP:

SIP is a peer-to-peer protocol and the peers in a session are known as User Agents-UAS. A user agent can work in the following roles: (i) User agent client: A client application which starts the SIP request. (ii) User Agent Server -A server application which establishes contact with the user at the time when a SIP request is received and that gives a response on behalf of the user. Normally, a SIP end point is able o work as both as UAC as also a UAS, but operates only as one or the other for every transaction. Whether the end point operates as a UAC or a UAS is dependent on the UA that started the request. (Overview of the Session Initiation Protocol)

The clients of the SIP are (i) Telephones which can act either as a UAS or UAC. Soft phones- PCs which have telephones abilities and CISCO SIP IP phones are able to start SIP requests and answer to requests. (ii) Gateways: These provide call control. Gateways are useful for a number of services, however, the most common being a translation function between SIP conferencing terminal points and other terminal types. This function comprises translation among transmission formats and among communication methods. Apart from that, the gateways translates between audio and video codes and does call setup and clearing on both the LAN sides and the switched-circuit network portion. SIP operates through servers which include (i) Proxy server, (ii) Redirect Server, (iii) Registrar Server.

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