Protocols used in voip phone communication

If talked about there are many protocols used for the voip phone communication but just two of them are the most common and used frequently. They are termed as H.323 and the (SIP) that means the Session Initiation Protocol.

1) H.323
The International Telecommunications Union (ITU) has specified H.323 as a protocol set that can lay a foundation for the IP supported real-time communications that can include data, audio and video. H.323 protocol can allow different configurations of data material, audio, video separately. For example the newer configurations can include data only, audio only, or video only or else audio & video, data and video, audio & data and video. But H.323 does not categorically prescribe the packet network or transport protocols.
This pattern of H.323 specifies four kinds of components as, Gateways, Terminals, Gatekeepers and Multi-point Control Units (MCU).

What are the H.323 Protocols?
1. Audio CODEC is an audio CODEC (encoder/ decoder) that digitizes the audio signal from its analog source that can be microphones before the transmission process begins through the H.323 terminal.

2. Video CODEC on the other hand, digitizes the video signaling process from its analog source that can be a video camera before transmitting it from the H.323 terminal.

3. Registration, Admission and Status (RAS) – RAS is the protocol between gateways, terminals and gatekeepers. It is used for gateway or terminal registration process, changing the bandwidth, controlling the admission status, disconnecting procedures.

4. H.225 Call Signaling Process is basically used for establishing a connection between two endpoints of H.323 that are its terminals and gateways.

5. H.245 Control Signaling process are when H.323 endpoints and gatekeepers are connected, it can exchange the end-to-end control messaging system that manages the entire operation of the H.323 endpoints.

6. Real-Time Transport Protocol (RTP) is the protocol used to define the end-to-end delivery system of real-time audio and video data. RTP is used to transfer data through the user datagram protocol (UDP).

7. Real-Time Transport Control Protocol (RTrnet CP) controls services, and functions to provide feedback of the data distribution.

What is the Session Initiation Protocol (SIP)
The Session Initiation Protocol (SIP) is a signaling process used to establish sessions of an IP network. A session is a simple point-to-point connection of a telephone or a multi-point conference system. The SIP has made possible the implementation of new system of IP telephony like web page dial system, Instant Messaging system, voice-enriching e-commerce and IP Centrex system. SIP is the preferable protocol in the VoIP telephony industry over other protocols. As with this telephony Internet application and web-based services have become easier.

The SIP Components
SIP has two components i.e. user agent and the network server. The user agent starts SIP requests while the SIP server is the networking device that handles the signals associated with several calls. It has a User Agent Client (UAC) and a User Agent Server (UAS) where the client element starts the call requests while the server element forms the contacts to the user when a SIP request is received.
The function of the SIP server is to give name resolution and find the user location. These messages can be passed to other servers through the next hop routing protocols. SIP servers functions in two different modes: stateful and stateless. The server in the stateful mode can remember the incoming requests and gives back responses to callers. While, a stateless mode does not keep information once the request has been sent. Other than these the SIP servers re-direct and forking i.e. a re-direct server can receive requests and then pass them to the next server. Through forking the server can split an incoming call and several locations can then ring at once.



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