![]() ![]() This works for audio calls, video conferences, and instant messages. The session begins with the link authentication. SIP typically utilizes Transmission Control Protocol (TCP) or UDP to make this happen. When a participant receives the invite, they can click on a link through the invite and join the call. This SIP message will look like an email. Data packets are sent between endpoints in the call, and those packets trigger a SIP request to the communication session. SIP servers are used to both start and end IP calls. That means that businesses can eliminate the costs that come with using a traditional PSTN phone system. With this process, companies using a SIP-enabled PBX can leverage the internet to send and receive calls. ![]() SIP is usually synonymous with SIP trunking. SIP became standardized by the Internet Engineering Task Force (IETF) through RFC 3261. It was developed by Mark Handley, Henning Schulzrinne, Eve Schooler, and Jonathan Rosenberg in 1996 and eventually became adopted as a way to simplify real-time calls. The Transport Protocol is another example of this. As we previously reported, SIP technology is the application layer protocol that initiates, maintains, and terminates IP calls and chats.
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