s martbridge Trading Solutions Pvt. Ltd.

The basic requirements of a Voip system are quite simple, real-world implementations are quite complex. Voip systems in widespread use today fall into three groups: systems using the H.323 protocol, systems using the SIP protocol, and systems that use proprietary protocols.

H.323 is a standard for teleconferencing that was developed by the International Telecommunications Union (ITU). It supports full multimedia audio, video and data transmission between groups of two or more participants, and it is designed to support large networks. H.323 is network-independent: it can be used over networks using transport protocols other than TCP/IP. H.323 is still a very important protocol, but it has fallen out of use for consumer Voip products due to the fact that it is difficult to make it work through firewalls that are designed to protect computers running many different applications. It is a system best suited to large organizations that possess the technical skills to overcome these problems. As a solution for a home or small office telephony system it is best avoided.

SIP (for Session Initiation Protocol) is an Internet Engineering Task Force (IETF) standard signalling protocol for teleconferencing, telephony, presence and event notification and instant messaging. It provides a mechanism for setting up and managing connections, but not for transporting the audio or video data. It is probably now the most widely used protocol for managing Internet telephony. Like all IETF protocols, SIP is defined in a number of RFCs (Request For Comments – the standards documents that define Internet standard protocols) principally RFC 3261.

A SIP-based Voip implementation may send the encoded voice data over the network in a number of ways. Most implementations use Real-time Transport Protocol (RTP), which is defined in RFC 3550. Both SIP and RTP are implemented on UDP which, as a connectionless protocol, can cause difficulties with certain types of routers and firewalls. Usable SIP phones therefore also need to use STUN (for Simple Traversal of UDP over NAT), a protocol defined in RFC 3489 that allows a client behind a NAT router to find out its external IP address and the type of NAT device. Thanks to STUN, setting up SIP-based Voip hardware or software behind a home or small office firewall should be a simple affair, but in practise it can still be problematic.

There is little that can be said about proprietary Voip implementations such as Skype or Voipcheap, because little or no technical detail about them is published. However, one advantage of proprietary implementations is that they can be designed with the principal objective of being install-and-run on an end-user computer without any fiddling with firewall settings being required. The disadvantage is that these products will not inter-operate with anything else. This may not seem such a bad thing to users of Skype, with its millions of members, but it might be inconvenient in the future