RTMP(Real Time Messaging Protocol) is an open Protocol developed by Adobe Systems for audio, video, and data transfer between Flash players and servers. It comes in three variants: 1) a plaintext protocol that works on top of TCP, using port 1935; 2) RTMPT is encapsulated in HTTP requests and traverses firewalls; 3) RTMPS is similar to RTMPT, but uses HTTPS connection; The RTMP protocol is used by Flash for object, video, and audio transmission. This protocol is built on top of TCP or polling HTTP. The RTMP protocol acts as a container for data packets, either in THE AMF format or in the FLV format. A single connection can transmit multiple network streams over different channels, where packets are transmitted in fixed size packets. RTSP(Real Time Streaming Protocol) defines how a one-to-many application can efficiently transmit multimedia data over an IP network. RTSP provides an extensible framework for data sources that can include both live data and existing stored data. The purpose of this protocol is to control multiple data sending connections, provide a way to select sending channels such as UDP, multicast UDP and TCP, and provide a method to select sending mechanisms based on RTP. RTSP has a similar syntax and operation to HTTP/1.1, but does not specifically emphasize time synchronization, so it is more tolerant of network latency. The caching function of the proxy server also applies to the RTSP, and because the RTSP has the redirection function, it can switch the servers that provide services according to the actual load to avoid the delay caused by excessive load concentration on the same server. Real-time Transport Protocol (RTP) RTP is a Transport layer Protocol for multimedia data streams. It describes the standard packet format for audio and video transmission over the Internet. RTP is commonly used in streaming media systems (in conjunction with RTCP), video conferencing, and one-touch communication systems (in conjunction with H.323 or SIP), making it the technical foundation of the IP telephony industry. RTP is based on UDP and is often used in conjunction with RTCP. RTP itself does not provide a delivery on time mechanism or other quality of service (QoS) guarantee. It relies on low-level services to implement this process. RTP does not guarantee transmission or prevent unordered transmission, and does not determine the reliability of the underlying network. It simply sends regardless of whether the transmission loses packets or whether the receiver has received packets. Sequential transmission is implemented in RTP. Serial numbers in RTP allow the receiver to reorganize the sender’s packet sequence, and serial numbers can also be used to determine the appropriate packet location. For example, in video decoding, sequential decoding is not required. 4. Real-time Transport Control Protocol (RTCP) RTCP is a supporting Protocol of RTP and provides out-of-channel Control for RTP media streams. RTCP and RTP work together to package and send multimedia data, periodically transferring control data between participants in a multimedia stream session. The main function of RTCP is to provide feedback on the quality of service (QoS) provided by RTP, collecting statistics on related media connections, such as bytes transferred, packets transferred, packets lost, one-way and two-way network latency, and so on. Network applications can take advantage of the information provided by RTCP to improve their quality of service, such as limiting traffic or switching to codecs with a lower compression ratio