IDM MCC600 Audio/Video Converged Communication Terminal Technical Explanation SIP Video Conference Technology

1. What is SIP in Video Conferencing? u SIP in video conferencing refers to the scenario where an application or website uses the Session Initiation Protocol (SIP) to facilitate real-time video conferences between two or more participants. SIP is an application-layer protocol that initiates, establishes, and terminates real-time data transmission between endpoints (such as devices, computers, applications, or local networks). Video conferencing requires real-time transmission of audio and video data, and SIP is one of the multiple protocols and technologies that enable this function. u Video conferencing over the Internet relies on real-time data transmission between all users. Since each participant uses different devices, browsers, and local networks, application-layer protocols such as SIP help software applications agree on the types of data participants will send, the servers that receive the data, and the speed at which the data is transmitted. u Many video applications, such as DingTalk Video Conference and Tencent Video, utilize proprietary APIs to provide the same functionality as SIP. 2. How Does SIP Video Conferencing Work? u SIP video conferencing establishes connections between multiple endpoints (such as devices, users, or applications) so that they can send video and audio data in real time. u The following is a detailed breakdown of how SIP supports video conferencing: 1. **The caller initiates the video conference**: When a user clicks the "Start Now" button to initiate a video conference, their device sends invitations to other participants through the local router network so that they can join the conference. Many software applications use proprietary protocols or APIs based on HTTPS, but many also use SIP to send invitation requests. 2. **The receiver receives and accepts the invitation**: SIP sends the request from the caller's server to the receiver's server, which then forwards it to the receiver. Once the receiver clicks the button or joins the link to attempt to join the conference, SIP sends this message back to the caller through both parties' servers. If multiple participants want to join, the servers and routers will repeat this process, inviting each participant and notifying the initial caller. 3. **SIP determines conference parameters**: Once the video call and participants are confirmed, SIP negotiates key information about the call details. SIP decides which codecs, media formats, and media communication protocols the call will use. Typically, SIP calls use the best-quality codec shared by the users. SIP also prefers the Real-time Transport Protocol (RTP) for video streaming and the User Datagram Protocol (UDP). These transport protocols support fast bidirectional data transmission, enabling users to experience real-time video conferencing with minimal jitter and latency. 4. **The media session proceeds**: Once SIP establishes the video connection, another protocol and API take over to actively transmit data during the session. For SIP video, this typically means that RTP facilitates the active media session throughout the entire session. 5. **SIP terminates the session**: When a user terminates or ends the session, their device sends a SIP request to the caller and any other participants. Once the other endpoints or devices receive this request, the call ends. 3. Advantages of SIP for Video Conferencing u Compared with traditional alternatives such as PSTN, H.323, and ISDN, SIP offers the following advantages for video conferencing: - **Fast**: SIP uses the User Datagram Protocol (UDP) as the data transmission layer, giving video conferencing minimal latency or lag. - **Secure**: SIP creates a secure connection between participants, encrypting all data transmitted between them. Therefore, SIP video conferencing is relatively secure. - **Supports a large number of users**: SIP video conferencing can support 1-to-1 video conferences, or up to thousands of video call participants. RTP and multicast can utilize different methods to transmit video and audio data among hundreds of participants with almost no latency. - **Scalability**: SIP not only supports a large number of users, but SIP-based video conferencing platforms can typically add and remove users easily, regardless of their location. Therefore, SIP conferencing supports scalability and flexible remote teams. - **Cloud hosting**: While some telephone systems and UCaaS platforms host their SIP solutions on-premises, SIP applications are typically hosted remotely by vendors. This saves costs associated with equipment maintenance and may allow your team to reduce IT staffing requirements. - **Accessible from multiple devices**: SIP video conferencing supports devices connected via the Internet: desktops, web browsers, tablets, and mobile applications. Participants can use any of these devices to connect to the Internet.