Ip Pbx And Its Uses
By San
The integrated voice features, for example unified messaging and "click-to-dial", became available with circuit-switched PBX systems. Complex coordination of voice and data networking environments are required. IP network level however, eliminates much of this complexity, attaining results which are lower costs from consolidating equipment and rapid integration with business applications. The requirements necessary for IP PBX systems to successfully integrate with a packet network are:

-Reliability to minimize the loss of voice packets and control network delay.
-Quality of Service (QoS) to ensure that voice packets are transmitted through each network element with the correct priority relative to other types of packets.

Real Time Protocol (RTP)
RTP uses a set of Internet Engineering Task Force (IETF) standard protocols for reliably transporting voice, video, and data among participating parties across a packet network. Comments requests include:

-The RTP protocol (RFC 2550), carrying voice that has been encoded using ITU G.711 or an alternative codec system. The protocol includes time stamps and sequence numbers, "jitter management".
-RTP for dual tone multi-frequency digits carries encoded tones used in interactive voice response (IVR) applications.
-RTP for redundant audio data specifies a mode of transporting same sequenced audio patterns to effectively reduce the number of packets transmitted.

H.323
H.323 - a set of related ITU recommendations that describe the architecture for multimedia communication over a packet-switched network. Key factors:

-H. 323, defining the multimedia infrastructure, including the role of devices, gateways and gate keepers.
-H. 245, specifying the call control processes for establishing and terminating a multimedia session, which includes the exchange of the capabilities supported by client

devices.
-H. 450, specifying a set of supplementary devices, like call transfer, hold, message waiting etc. This protocol is useful communication between clients without the aid of a switch.

Session Initiation Protocol (SIP)
SIP is an IETF standard protocol helpful for initiating an interactive user session that carries multimedia subjects. The SIP architecture comes with a method for delivering connections for users at any location where he/she is registered. In contrast to the H.323, SIP uses an extensible text-based format, similar to HTTP. SIP, being the basis for multi applicable programs including instant messaging and gaming. It can function as a standalone call control protocol or together with H. 323.

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Article contributed by Phil Lam who is a consultant at Lantone Communications. Led by a team of experienced IT developers, Lantone Communications is one of the leadingVoIP Provider in Singapore. Please visit their official website for the latest information on VoIP. This article may be reprinted in its original form as long as the resource box is left intact and the links live and the article is not to be modified in any way.