The SIP line, or communication line based on the Session Initiation Protocol, is a core component of modern Voice over IP (Voice over IP) services. The working principle of the SIP line involves the interaction of several key steps and components, which is explained in detail below.
SIP is a text-based protocol for establishing, modifying, and terminating multi-session. It uses UDP (User Datagram Protocol) or TCP (Transmission Control Protocol) as the transport layer protocol and communicates via messaging. A SIP message consists of a request that is sent by the client to the server and a response that is sent back to the client by the server.
Make a request: When a user wishes to establish a communication session (e.g., making a call), the SIP client (e.g., a soft application) constructs a SIP request message (e.ginviterequest), which contains the destination address (i.e., the called party's SIP URI) and a description of the session (e.g., type, codec, etc.).
Routing and **: The SIP request message is sent over the network to a SIP** server (such as a SIP proxy). The server is responsible for routing and sending the request to the SIP server where the destination address is located.
The called party responds: After receiving the request, the IP server of the called party notifies the called user. If the called party accepts the call, it sends one through the SIP server200 okRespond to the message back to the caller.
The session is established: Once the caller receives it200 okresponse, the session is considered established. Subsequently, a Real-Time Transport Protocol (RTP) session is established between two parties for the actual transmission of voice, **, or data streams.
Session Transfer: During the session, you can use SIP'stransferRequest to transfer a session from one device to another.
Session persistence: Can be used if you need to temporarily interrupt the session (e.g. by setting ** aside).holdrequest to pause the session.
Session termination: When a party wishes to end a session, it sends onebyerequest to terminate the session. When the other party receives the request, it will be sent200 okresponse to confirm the end of the session.
SIP is also responsible for negotiation during the process of establishing a session. This involves choosing an audio codec, codec, and other parameters that are acceptable to both parties. This is usually done through the Session Description Protocol (SDP), which is included in the SIP request and response messages.
SIP client: Typically a soft application or hardware device on a user's device that is used to make and receive SIP requests and responses.
SIP server: includes SIP** servers, registration servers, and redirect servers. They are responsible for handling SIP requests, performing routing and operations, and maintaining user registration information.
rtp/rtcp: For actual streaming and control. RTP is responsible for transmitting audio, **, or data streams, while RTCP is responsible for providing flow control and quality monitoring.
Network quality: VoIP communication has high requirements for network quality, and problems such as network delay, packet loss, and jitter may affect call quality.
NAT and firewalls: Certain network configurations, such as NAT and firewalls, may prevent the transmission of SIP signals and require proper configuration and ports**.
Security: While SIPs support encrypted communications (such as TLS), they can be exposed to security risks such as man-in-the-middle attacks or eavesdropping if not configured correctly.
Interoperability: Different devices and systems may use different SIP implementations and codecs, which can lead to interoperability issues.
In general, the working principle of SIP lines involves the interaction of multiple components and protocols, including SIP, RTP RTCP, and SDP. By working together with these protocols, SIP lines enable flexible, efficient, and cost-effective multi** communications. However, in practice, there are challenges to consider such as network quality, security, and interoperability.