SIP creates a VoIP system that connects you to the phone networks. On the contrary, in VoIP a single system is responsible for controlling all of the functions.
The speech is digitised and encoded with the help of the codec device installed in the IP phones. To make the voice calls locally, the LAN is used. Referring to the below-given diagram, the IP telephone system is linked to the wide area IP network and LAN too. Later, the advent of IP emerged the concept of packet switching which makes communication possible by dividing the data into small sized self-reaching packets (because it contains the destination address).Earlier the telephone system named as PSTN (Public Switched Telephone Network) work on the circuit switching, where the resources are engaged until the call is ended.At last, an isochronous network of the IP network. Secondly, it requires a mechanism to set and end the calls. Priorly, it needs a protocol like RTP to accurately transfer the digitised signal along the IP network. There are three components necessarily required for achieving VoIP (Voice over IP). The alternative name for the VoIP is the IP telephony. VoIP (Voice over IP) is the utilisation of IP for enabling the telephone service. It can also place a call from an IP enabled equipment to an ordinary telephone. SIP can also provide features such as call waiting, call screening, authentication and encryption. The SDP (Session Description Protocol) is used to provide the information about the call. B2B(Business-to-Business) User Agents – Includes two user agent capable of transferring and modifying the SIP messages.Gateways – Gateway is nothing but a user agent which can connect to other networks, for example, PSTN.Various Servers – There are several types of servers employed in the SIP protocol, such as proxy, location, registrar, redirect.User Agents – Client and server fall under the user agent category, in which the client creates requests while server receives the requests and generate responses.There are generally four components of Session Initiation Protocol, which are discussed below. The session is terminated by sending a BYE method to either party.Then the caller generates an ACK message to culminate the protocol and confirm the receipt of the 200 (OK) message.After that, the connection is completed using the three-way handshaking protocol.The callee can also send OPTIONS method to query the host about its capabilities and other information.
SIP DEFINITION IT CODE
In case callee accepts the request, the reply code is sent to the caller.The proxy server placed between the caller and callee, examine the structure of the message body comprised of media type, format and caller’s capabilities.Firstly, the SIP caller generates a request in which an invite message is sent to the callee.
The entire process of the SIP protocol is explained in the steps below: In simpler terms, SIP works in a similar way as the HTTP works, where a request and response model is followed. The SIP is the control protocol of the application layer responsible for setting, manipulating and ending the multimedia sessions with one or more users, the multimedia session can contain any sort of multimedia, including data, voice, video, image, etcetera. SIP (Session Initiation Protocol) is a group of rules that govern the multimedia exchange over the internet.
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