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Overview of Voice-over-IP 1. Introduction: The term “Voice-over-IP” is one of the buzzwords of today’s IT industry and as well as of Telecommunication industry. The reason for its popularity can be viewed as its lucrative market potential and its versatility of application. The applications of VoIP ranges from personal phone-to-phone, PC-to-phone, PC-to-PC calls to corporate applications like Call Center, toll free numbers and many others. In this paper we will be defining VoIP, look at the causes for its evolution, and see how it actually works, i.e. analyzing the nuts and bolts of VoIP. 2. What is VoIP? Voice-over-IP (VoIP) implementations enable users to carry voice traffic (for example, telephone calls and faxes) over an IP network. [VoIP 2003] Because of its superior cost/benefit, VoIP provides a competitive threat to providers of traditional telephone services. But in terms of quality of service (QoS), VoIP cannot yet be the substitute to the traditional telephone services. 3. How does VoIP work? Unlike circuit-switched networks, there is no dedicated path between communicating end systems over the Internet. The Internet uses a technique known as packet switching that allows multiple communicating end systems to share a path, or parts of path, at the same time. [Kurose 2003] In VoIP, the digital signal processor (DSP) segments the voice signal into frames and stores them in voice packets. These voice packets are transported using IP in compliance with one of the specifications for transmitting multimedia (voice, video, fax and data) across a network. 4. Standards for VoIP The standards bodies, such as the IETF and ITU, have been busy for many years (and continue to be busy) at hammering out standards for VoIP applications. In this section we will briefly examine RTP, SIP and H.323, three of the most widespread implementing sets of standards among the existing ones. 4.1 RTP RTP (Real Time Protocol) is defined in RFC 1889, as a standard of real-time interactive application. RTP can be used for transporting common formats such as PCM, GSM, and MP3 and MPEG and H.263 for video. It can also be used for transporting proprietary sound and video formats. Today, RTP enjoys widespread implementation in hundreds of products and research prototypes. It is also complementary for other real-time interactive protocols, including SIP and H.323. [Kurose 2003] RTP typically runs on top of UDP. The sending side encapsulates the media chunk within an RTP packet, then encapsulates the packet in a UDP segment, and then hands the segment to IP. The receiving side extracts the RTP packet from the UDP segment, then extracts the media chunk from RTP packet, and then passes the chunk to media player for decoding and rendering. 4.2 SIP Among the most promising protocols in this direction is the Session Initiation Protocol (SIP), defined in [RFC 3261]. SIP is a light weight protocol that does the following: Ř It provides mechanisms for establishing calls between a caller and a callee over an IP network.
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