When conducting live broadcasts and video calls, the smooth experience cannot be achieved without the support of multiple network transmission protocols. Although the names of these protocols sound very technical, the choice of these protocols will directly affect whether lag occurs during viewing, and will also directly affect the level of clarity during viewing.
Characteristics and applications of RTMP protocol
The RTMP protocol was developed by Adobe specifically for Flash players. Its main purpose is the real-time transmission of audio and video data. It requires the establishment of a continuous and stable connection between the client and the server to achieve the purpose of continuously pushing data streams.
This protocol was once widely used in PC-side web live broadcasts because early browsers generally supported Flash plug-ins. However, because mobile devices basically do not support Flash, RTMP is currently used more in the live streaming process, which is the process in which the anchor transmits the image from the encoding software to the live broadcast server.
Security enhancement of RTMPE protocol
RTMPE adds a complete encryption mechanism to the standard RTMP protocol. This means that the audio and video data content during the transmission process will be encrypted to avoid being stolen or tampered with by third parties during transmission.
For example, there are some corporate live broadcasts or paid courses. Such companies have high requirements for content security. They may use the RTMPE protocol to ensure that the content will not be illegally recorded or disseminated illegally. This protocol makes up for the security shortcomings of the RTMP protocol, which was originally designed to be relatively simple.
Control framework of RTSP protocol
The RTSP protocol does not directly transmit data streams. It is like a remote controller, responsible for establishing and controlling media sessions. It defines instructions such as play, pause, stop, etc., allowing the client to control the media playback behavior of the server.
Because the control flow and data flow are separated from each other, the RTSP server can flexibly choose to use TCP or UDP protocol to transmit actual audio and video data according to the network conditions. This design still has applications in some network monitoring and IPTV systems.
Real-time transmission of RTP protocol
The protocol actually responsible for transmitting real-time audio and video data packets is the RTP protocol. It is usually transmitted via UDP. The sender will mark each data packet with a sequence number and timestamp in order, and then send it out quickly without waiting for confirmation.
Using this "send it and forget about it" approach gives up part of the reliability, but in exchange for extremely low latency, which can be accurate to the millisecond level, which is extremely critical and important for real-time calls. The receiving end will reassemble and synchronize the audio and video based on the sequence number and timestamp.
Collaborative feedback of RTCP protocol
RTCP is a companion protocol to RTP, which regularly transmits control information between communication participants. This protocol does not transmit media data, but sends packet statistics reports such as packet loss rate, delay and jitter.
The sender can dynamically adjust the encoding quality or the sending rate based on the feedback information. For example, when network packet loss is severe, you can proactively reduce the video bit rate in exchange for a more consistent picture, thereby maintaining usable services as much as possible under harsh network conditions.
The role of HTTP and FLV in streaming media
Relying on HTTP streaming media technology, such as HLS or DASH, the entire video file will be cut into many small fragments, which are downloaded and played in order. This mode is compatible with all devices and browsers that support HTTP, and is very suitable for on-demand and live broadcasts with relatively high delays.
The structure of the FLV format shows simple characteristics. Its stream data can be transmitted with the help of the HTTP protocol, and then form an HTTP-FLV stream. Its support on mobile browsers is limited, but it has shown stable performance when used as a live broadcast protocol in mobile APPs. Many domestic live broadcast platforms have adopted this solution.
In various scenarios such as live broadcasts, video conferencing, and online education, which aspect do you value most, transmission speed, image quality, or compatibility? You are welcome to share your views in the comment area. If you find this article helpful, please like it to support it.
