The Session Initiation Protocol (SIP) is an application-layer control protocol used to create, modify, and terminate sessions between participants, including Internet telephony calls, multimedia distribution, and conferences. SIP enables users to discover each other, agree on session parameters, and manage sessions. It uses proxy servers to route requests, authenticate users, and provide call-routing policies. SIP supports various transport protocols and includes features like registration, which allows users to upload their current locations for use by proxy servers. SIP messages include requests, responses, and header fields that provide information about the session. The protocol defines a set of methods, such as INVITE for initiating a session, BYE for terminating it, and CANCEL for canceling a request. SIP also includes a set of response codes to indicate the status of requests, ranging from provisional (1xx) to final (2xx, 3xx, 4xx, 5xx, 6xx) responses. SIP is structured as a layered protocol, with layers for syntax and encoding, transport, and transaction handling. The transaction layer manages requests and responses, while the transport layer handles the actual communication over networks. SIP also includes security mechanisms, such as authentication, integrity protection, and encryption, to ensure secure communication. SIP works with other protocols like RTP for real-time data, RTSP for streaming media, and SDP for session descriptions. It is not a vertically integrated system but a component that can be used with other IETF protocols to build a complete multimedia architecture. SIP is used for session setup, modification, and termination, and it supports features like call control, media negotiation, and session management. The protocol is designed to be flexible and scalable, allowing for various use cases such as voice, video, and text messaging.The Session Initiation Protocol (SIP) is an application-layer control protocol used to create, modify, and terminate sessions between participants, including Internet telephony calls, multimedia distribution, and conferences. SIP enables users to discover each other, agree on session parameters, and manage sessions. It uses proxy servers to route requests, authenticate users, and provide call-routing policies. SIP supports various transport protocols and includes features like registration, which allows users to upload their current locations for use by proxy servers. SIP messages include requests, responses, and header fields that provide information about the session. The protocol defines a set of methods, such as INVITE for initiating a session, BYE for terminating it, and CANCEL for canceling a request. SIP also includes a set of response codes to indicate the status of requests, ranging from provisional (1xx) to final (2xx, 3xx, 4xx, 5xx, 6xx) responses. SIP is structured as a layered protocol, with layers for syntax and encoding, transport, and transaction handling. The transaction layer manages requests and responses, while the transport layer handles the actual communication over networks. SIP also includes security mechanisms, such as authentication, integrity protection, and encryption, to ensure secure communication. SIP works with other protocols like RTP for real-time data, RTSP for streaming media, and SDP for session descriptions. It is not a vertically integrated system but a component that can be used with other IETF protocols to build a complete multimedia architecture. SIP is used for session setup, modification, and termination, and it supports features like call control, media negotiation, and session management. The protocol is designed to be flexible and scalable, allowing for various use cases such as voice, video, and text messaging.