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Streaming overview:
The so-called streaming media refers to the media format played on the Internet by means of streaming transmission.
Streaming media is also called streaming media. It refers to businesses that use a video delivery server to send out programs as data packets and deliver them to the network.
After the user decompresses the data through the decompression device, the program will be displayed as before transmission.
Streaming media transmits audio, video and multimedia files in the form of streaming over the network.
Streaming media file format is a media format that supports streaming and playback.
The streaming method is to divide multimedia files such as video and audio into compressed packages through a special compression method.
Continuous and real-time transmission from the server to the user's computer. In systems that use streaming transmission, users don’t have to wait for the entire file like non-streaming playback.
The content can be seen after all downloads are completed, but only a few seconds or tens of seconds of startup delay can be used on the user’s computer
The corresponding player plays the compressed video or audio and other streaming media files, and the remaining part will continue to be downloaded until the playback is complete.
1. RTP: (Real-time Transport Protocol)
It is a transport layer protocol for multimedia data streaming on the Internet. The RTP protocol and the RTP control protocol RTCP are used together,
And it is built on the UDP protocol.
RTP is not like http and ftp which can download the entire movie file completely. It sends data on the network at a fixed data rate. The client also watches the movie file at this speed.
After the movie screen has been played, it cannot be played repeatedly, unless the data is requested from the server again.
2. RTCP: Real-time Transport Control Protocol or RTP Control Protocol or RTCP for short)
The real-time transmission control protocol is a sister protocol of the real-time transmission protocol (RTP).
Note: --: RTP protocol and RTP control protocol (RTCP) are used together, and it is based on UDP protocol (usually used for video conferencing)
3. RTSP: (Real Time Streaming Protocol)
Real-time streaming media session protocol, SDP (session description protocol), RTP (real-time transport protocol).
It is a multimedia streaming protocol used to control sound or video. RTSP provides an extensible framework that makes it possible to control and on-demand real-time data, such as audio and video.
The media data uses rtp and rtcp protocols. Generally use udp as the transport layer. Suitable for IPTV scenes. Data sources include live data and data stored in clips. The purpose of this protocol is to control multiple data transmission connections, to provide a way to select transmission channels, such as UDP, multicast UDP and TCP, and to provide methods for selecting a transmission mechanism based on RTP. The network communication protocol used during transmission is not within the scope of its definition. The server can choose to use TCP or UDP to transmit the streaming content, which is more tolerant of network delays.
--->: The biggest difference between RTSP and RTP is that: RTSP is a two-way real-time data transmission protocol, which allows the client to send requests to the server, such as playback, fast forward, and reverse operations. when
Of course, RTSP can transmit data based on RTP, and can also choose TCP, UDP, multicast UDP and other channels to send data, which has good scalability. It is similar to the http protocol
Network application layer protocol.
4. WebRTC:
The web side implements the streaming media protocol. When Google first launched WebRTC, the giants were either sitting on the sidelines or resisting it. Use RTP protocol transmission.
5. RTMP (Real Time Messaging Protocol)
A set of live video protocols developed by Macromedia is now owned by Adobe. Like HLS, it can be applied to live video, and it will not be lost based on TCP.
// The difference is that RTMP cannot be played in the iOS browser based on flash, but the real-time performance is better than HLS.
The real-time messaging protocol is an open protocol developed by Adobe Systems for audio, video, and data transmission between Flash players and servers.
// In the iOS code, RTMP is generally used to push the stream. You can use the third-party library librtmp-iOS to push the stream. librtmp encapsulates some core APIs for users to call
The RTMP protocol also requires the client and server to establish an RTMP Connection through a "handshake", and then transmit control information on the Connection. The RTMP protocol will format the data during transmission. In the actual transmission, in order to better achieve multiplexing, sub-packaging, and fairness of information, the sender will divide the Message into Chunks with Message ID, each Chunk May be a single message,
It may also be part of the Message. The receiving end will restore the chunk to a complete Message according to the length of the data contained in the Chunk, the length of the message id and the length of the message, so as to realize the sending and receiving of information.
6. HLS: HTTP Live Streaming (HLS)
It is an HTTP-based streaming media transmission protocol implemented by Apple Inc., which can realize live and on-demand streaming media. It is mainly used in the iOS system and provides audio and video live and on-demand solutions for iOS devices (such as iPhone and iPad). HLS on-demand is basically a common segmented HTTP on-demand. The difference is that its segments are very small. Compared with common streaming media live broadcast protocols, such as RTMP, RTSP, MMS, etc., the biggest difference in HLS live broadcast is that what the live broadcast client obtains is not a complete data stream.
The HLS protocol stores the live data stream as continuous, short-duration media files (MPEG-TS format) on the server side, and the client continuously downloads and plays these small files, because the server side will always update the latest live broadcast The data generates new small files, so as long as the client continuously plays the files obtained from the server in sequence, the live broadcast is realized. It can be seen that, basically, it can be considered that HLS is a technical way of >>on-demand to realize live broadcast <<. Since the data is transmitted through the HTTP protocol, there is no need to consider firewalls or proxies at all.
In addition, the duration of the segmented file is very short, and the client can quickly select and switch the bit rate to adapt to the playback under different bandwidth conditions. However, this technical characteristic of HLS determines its
The delay will always be higher than the ordinary live streaming protocol.
// Both iOS and Android naturally support this protocol, the configuration is simple, just use the video tag directly
***VLS: It is a kind of streaming server, which is specially used to solve various problems of streaming. It also has some characteristics of VLC. As a server, videolan can output http, rtp, rtsp streams.
In principle, RTSP, RTMP, and HTTP can all be used for live broadcast and on-demand broadcasting, but generally RTSP and RTMP are used for live broadcasting, and HTTP is used for on-demand broadcasting. What we chose is the RTMP protocol.
Various protocol delays and their reasons
rtmp and httpflv: The data of these two protocols are roughly the same, so the reasons for the delay are the same. It stands to reason that tcp streaming live broadcasts should have extremely low latency. Why do rtmp and httpflv still have latency? The reason is that on h264, rtmp and httpflv are flv tags that are transmitted. The data of video tag is usually h264 data. H264 decoding has an IBP. I is a key frame, which is a complete image. You must have an I before decoding. The latter BP, BP frames can be as few as you want, but I frames cannot be less, so I frames must be transmitted second in flv tag transmission (the first is h264spspps), but I frames are not often present in h264 streams , There is an I frame after a period of time. This period of time is commonly known as GOP. When encoding, the GOP is set very short. When the client connects, the server will find the nearest I frame in the stream at the fastest speed, and send it from the I frame. Live data, but when the GOP is very long, the I frame interval is very long, or wait for the next I frame to start sending data to the new connection, or find the nearest I frame in the buffer to start sending, here is the rtmp and hls protocol delay The key point is that on major CDN platforms, it is called "rtmp second opening technology". The principle is to decode the push data twice and set a small gop. In general, gop is set to 1s. Regardless of the delay of the network transmission link, the maximum data delay is 1s. Fortunately, the I frame is 0 delay!
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