Over the next few years the use of bandwidth will increase and the industry is looking for a solution for the problems, which will arise by the growing use of bandwidth. The best solution so far is ATM (asynchronous transfer mode) the switching fabric designed to handle voice, data and video. This is a good step on the way, but there will be still problems with network congestion and packet loss. Further on the industry must also tackle the problems of network reliability and sound quality. This can be done by the gradual adoption of standards. These standards will focus on three central elements; the audio codec format, transport protocols and directory services.
The H.323 protocol:
In May 1996, the International Telecommunications Union (ITU) ratified the H.323 specifications (an adaptation of H.320), which defines how voice, data and video traffic will be transported over IP based local area network. H.323 is a set of recommendations and is based on the transport protocol RTP. The RTP is a new protocol layer for real-time applications. The limitation of RTP is that the protocol has no mechanisms for ensuring the on-time delivery of traffic signals or for recovering lost packets. RTP also does not address the so-called Quality of Service (QoS) issue related to guaranteed bandwidth availability for specific applications. At the moment there is a draft signaling-protocol standard that can strengthen the Internet ability to handle real-time traffic reliably. This makes it possible to setup dedicated end-to-end transport paths for specific sessions much like the circuit-switched PSTN does. This protocol is called the resource reservation protocol (RSVP)
H.323 specifies five phases for call completion:
- Call Initiation
- Endpoint capability exchanged
- Establish the call
- Call services (such as bandwidth utilization)
- Call termination.
The H.323 standard describes the roles that must be played for successful interoperability of the gateway, terminal devices gatekeeper and multipoint control unit. While H.323 has been widely accepted as the standard protocol for voice-over-IP networking, another protocol called Session Initiation Protocol, or SIP, has been gaining popularity as an alternative. SIP, a client-server protocol, was originally created to control multimedia calls over the internet as is similar to HTTP.