Voice over ip and its uses

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Voice Over IP

Voice over IP also called VoIP, known as internet telephony. Where traffic flows over an IP based network without compromising Quality of service (QoS) and cost. These efforts has started from 1990's and advanced rapidly as an alternative to Public Switched Telephone Networks (PSTN) due to cost effectiveness. VoIP has come up with the services like voice mails, video conferencing, skype etc. [1].

VoIP Protocols

The basic protocols used in VoIP are Transmission Control Protocol (TCP) and Internet Protocol (IP) and other related protocols. After using an internet as a medium, It has also used TCP/IP protocol suit. VoIP basically runs on top of the transport layer, starts from top to the bottom of the encapsulation process. Any data link or physical layer can be used. In transport layer, large amount of setup delays introduced by TCP, which can makes it incompetent for voice communication. On the other hand the use of UDP provide faster data exchange but without consistent data delivery. Alternatively Real-time Transport protocol (RTP) and Real User Datagram Protocol (RUDP) are the first choice of use in VoIP for providing efficient delivery and media on demand [2].

RTP

The RTP standard defines a packet structure for use in real-time applications, it provides the fields for timestamps and sequence numbers for synchronization, Sequence number field helps in reordering the packets, Packet loss detection. On the other hand Timestamp helps the receiver dealing with jitter, jitter estimation, synchronization. Other fields depend on PT field value. RTP provides tools for reproducing the contents but does not provide functionality [2]. This purpose is fulfilled by RTCP which provide additional information about data exchange and network performance. It uses different port no. then RTP stream [3], It also provides gateway support and group identification to allow group teleconferencing in near real time. RTCP doesn't sends a payload information with application data but sends statistical information, necessary for the application program to provide feedback on VoIP quality[2].

The downside of using RTP is the additional overhead because the IP/UDP/RTP header is 40bytes long. which is one half of the header size[2].

RUDP

RUDP is used alternatively, because it provides some consistency and Survivability to UDP, this consistency is achieved by sending more than one copies of packet to the destination , as one of them would make to its destination on time. It is able to provide one-by-one delivery of the packet in a reliable way. RUDP is bandwidth consuming but even it is used where reliability is the major concern, no matter double or triple bandwidth is being consumed [2].

VoIP Signalling

SIP AND H.323

VoIP consist of two parts: media transmission and signalling. Signalling process involves establishing calls, authentication users, controlling status of call, setting up route, call terminating. This is the mostly used protocol now a days[6] as compare to H323. H323 based on previous PSTN architectures. SIP is using the HTTP protocol, with request response model. It focuses directly on internet architecture. In fact it re-uses the many Internet protocols and other possible elements, when new things are needed, it tries to keep them at minimum and simple.

SIP was developed by Henning Schulzme (Columbia University) and Mark Handley (UCL) in 1996. By then it has been used increasingly with success and then got accepted in VoIP community in November 2000. 3GPP was adopted as signalling protocol and a permanent element for the new IP Multimedia Subsystem (IMS) [4][5].

SIP basically, provide a simple and lightweight means of creating and ending the connections for real- time interactive communication over IP networks. Specially for video, voice, gaming or even application sharing. Its main focus is on call control but not on the QoS, management, mobility, media synchronization and/or mixing. SIP supports an application that has more complex functionality in terms of implementing video downloading service through internet.

SIP packets are called messages. SIP request message carries a method specifying the request type, and the corresponding response carry status code showing the answer. The original SIP has six methods which make its process simple. A large no. of status codes built besides the lines of HTTP, Each consists of 3-digit code, First indicates the general kind of answer, and last two gives more concrete information, like 183, 1 information response, 83- session in progress, 400, 4- request failure, 00- bad request, in case everything working properly, is 200 OK.

VoIP signalling will not be discussed so much because the emphasis in on the performance and the quality of service.

Voice CODECS

VoIP services are increasingly being deployed over various types of packet switched IP networks that include internet, Enterprise LAN and wireless networks. The main element of any VoIP software is the codec which determines the voice speech are encoded into packets to send over the network. Now a days, various range of codec are available to match the performance requirements and the network environment [7].

Real voice quality depends on the codec's performance in terms of accuracy in reproducing the speech at receiver end. Its not only the performance but also the techniques followed by codec in process of voice communication. Mainly three steps are followed during voice transmission, which are as follows. Waveform is sampled at regular intervals by codec and generates a value for each sample. Samples are typically taken 8000 times/s (8 KHz sampling rate) or 16000 times/s. Then, the sampled values are quantized in order to record values into discrete-finite values that will be represented using bits. The coding step deals with gathering the samples for a set period of time and encode into a set of bits, to form a data frame that is transmitted over the network. On the whole process this is the last step which diagnosis the effectiveness of the codec in terms of the bandwidth requirement. The main objective of the speech coding during the process, is to reduce the bit rate in digital representation without any compromise in loosing the signal quality. Along with the above consideration the speech coding process must also have less complexity and less packetization delay. Complexity means time and space complexity of the coding algorithm. The packetization delay means maximizing maximum waiting time of voice sample until voice frame is created that contains the sample. International Telecommunication Union (ITU) has standardised these codec's for PSTN networks as G.711 and G.72x Series of codec's. PSTN lines offered low bandwidth that encouraged the design goal of low bit rate voice communication. As a result, those codec's were fall under the class of narrowband codec's which were developed for PSTN. These codec's are still popular in providing voice over PSTN but also used in voice over wireless cellular networks. [9-11]

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