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1. RTP/RTCP protocol
RTP protocol
The full name of RTP is Real-time Transport Protocol (Real-time Transport Protocol).
It is a standard proposed by IETF (The Internet Engineering Task Force), and the corresponding RFC document is RFC3550 (RFC1889 is an expired version). RFC3550 not only defines RTP, but also defines the supporting related protocol RTCP (Real-time Transport Control Protocol, that is, Real-time Transport Control Protocol). RTP is used to provide end-to-end real-time transmission services for a variety of multimedia data that need to be transmitted in real time, such as voice, image, and fax on the IP network. RTP provides time information and stream synchronization for end-to-end real-time transmission on the Internet, but does not guarantee service quality, which is provided by RTCP.
RTP application environment
(1) Simple multicast audio conference. Voice communication is realized through a multicast address and a pair of ports. One is for audio data (RTP) and the other is for control packets (RTCP).
(2) Audio and video conferencing. If both audio and video conferences are used in a conference, these two media will be transmitted in different RTP sessions, and each session will use a different transmission address (IP address + port). If a user uses two sessions at the same time, the RTCP packet corresponding to each session uses the canonical name CNAME (Canonical Name). Participants can obtain the associated audio and video according to the CNAME in the RTCP packet, and then synchronize the audio and video according to the timing information (Network time protocol) in the RTCP packet.
(3) Translator and mixer. The translator and mixer are both RTP-level relay systems. Translators are used in user areas that cannot be directly reached through IP multicast, such as a firewall between the sender and the receiver. When the audio encoding format that participants can receive is different, for example, if a participant connects to a high-speed conference through a low-speed link, a mixer is used. Before entering the network where the audio data format needs to be changed, the mixer reconstructs the audio packets from one source or multiple sources, merges the reconstructed multiple audios, and encodes them with another audio encoding. Forward this new RTP packet. All data packets from a mixer should be identified by the mixer as their synchronization source (SSRC, see RTP encapsulation), and the talker can be confirmed through the contribution source list (CSRC table, see RTP encapsulation).
RTCP protocol
Real-time Control Protocol (RTCP) and RTP are jointly defined in RFC 1889 proposed in 1996. It is a control protocol that works with RTP. RTCP runs on the low-level protocol alone, and the low-level protocol provides multiplexing of data and control packets. During the RTP session, each session participant periodically sends RTCP control packets to all other participants. For RTP sessions or broadcasts, a single multi-target broadcast address is usually used. All RTP and RTCP packets belonging to this session use this multi-target broadcast address. RTP packets and RTCP packets can be distinguished by using different port numbers. .
is a sister protocol of Real Time Transport Protocol (RTP). RTCP provides out-of-band control for RTP media streams. RTCP itself does not transmit data, but cooperates with RTP to package and send multimedia data. RTCP periodically transmits control data between participants in a streaming multimedia session. The main function of RTCP is to provide feedback on the Quality of Service provided by RTP.
RTCP performs the following four functions:
(1) Mainly to provide feedback on the quality of data release. RTCP is a part of the RTP transmission protocol and is related to the flow and congestion control of other transmission protocols. Feedback has a direct effect on adaptive encoding control, but IP multicast experience shows that receiving feedback from the sender is crucial for diagnosing transmission errors. Sending and receiving feedback reports to all participants allows problem observers to estimate whether those problems are local or global. Publishing mechanisms such as IP multicasting make it possible for groups such as network service providers to receive feedback and act as third-party monitors to diagnose network problems. The feedback function is performed by the RTCP sender and receiver reports.
(2) RTCP carries an RTP source persistent transport layer identification called a canonical name (CNAME). If a conflict is found, or the program is restarted, since the SSRC identity can be changed, the recipient needs CNAME to track the participant. The receiver also needs the CNAME to contact several data streams given in the relevant RTP connection.
(3) The first two functions require all participants to send RTCP packets. Therefore, in order for RTP to expand to large-scale quantities, the rate must be controlled. Let each participant send control packets to other participants, which increases the number of independent observation participants. This number is used to calculate the rate at which packets are sent.
(4) The optional function is to transmit minimum connection control information, such as identifying participants. Most likely to be used in "loose control" connections, where participants are free to enter or leave without member control or parameter coordination. RTCP acts as a convenient channel to all participants, but does not have to support all control communication requirements of the application.
When RTP is used in IP multicast, the first three functions are necessary and recommended for all situations. RTP application designers must avoid using mechanisms that only work in unicast mode, which will result in the inability to scale.
2. The relationship between RTP/RTCP and other protocols
Streaming media architecture diagram
The relationship between RTP protocol and other protocols
RTP, TCP, and UDP are all transport layer protocols; RTP can also be considered to be between the application layer and the transport layer
As can be seen from the figure, RTP is divided into the transport layer, which is built on UDP. Like the UDP protocol, in order to realize its real-time transmission function, RTP also has a fixed encapsulation form. RTP is used to provide time information and stream synchronization for end-to-end real-time transmission, but it does not guarantee the quality of service. The quality of service is provided by RTCP.
3. RTMP protocol
RTMP (Real Time Messaging Protocol) real-time messaging protocol is an open protocol developed by Adobe Systems for audio, video and data transmission between Flash players and servers.
It has three variants:
1) A plaintext protocol working on TCP, using port 1935;
2) RTMPT is encapsulated in the HTTP request and can traverse the firewall;
3) RTMPS is similar to RTMPT, but uses HTTPS connection;
The RTMP protocol is used by Flash for the transmission of objects, video, and audio. This protocol is based on the TCP protocol or the polling HTTP protocol;
The RTMP protocol is like a container used to hold data packets. These data can be data in AMF format or video/audio data in FLV;
A single connection can transmit multiple network streams through different channels. The packets in these channels are all transmitted in fixed-size packets;
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