RTP/RTCP、TCP、UDP、RTMP、RTSP(2)

Reference document RFC2326
The Real Time Streaming Protocol (Real Time Streaming Protocol) is a multimedia streaming protocol used to control sound or video, and allows simultaneous multiple streaming demand control. The network communication protocol used during transmission is not within its defined range. The server side You can choose to use TCP or UDP to transmit streaming content. Its syntax and operation are similar to HTTP 1.1, but time synchronization is not particularly emphasized, so it can tolerate network delays. The aforementioned multi-streaming demand control (Multicast) mentioned earlier can not only reduce the network usage on the server side, but also support multi-party video conferences (Video Conference). Because it works similarly to HTTP1.1, the cache function "Cache" of the proxy server "Proxy" is also applicable to RTSP, and because RTSP has a redirection function, the server that provides the service can be switched according to the actual load situation to Avoid excessive load concentrated on the same server and cause delay.

　　 was jointly proposed by Real Networks and Netscape. The protocol defines how one-to-many applications can effectively transmit multimedia data through an IP network. RTSP provides an extensible framework that makes it possible to control and on-demand real-time data, such as audio and video. 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.

RTP transmits audio/video data. If it is PLAY, the server sends it to the client. If it is RECORD, it can be sent to the server by the client. The entire RTP protocol consists of two closely related parts: RTP data protocol and RTP control protocol (ie RTCP) ；

　　 RTCP: RTCP includes Sender Report and Receiver Report, used for audio/video synchronization and other purposes, and is a control protocol;

　　RTSP requests mainly include DESCRIBE, SETUP, PLAY, PAUSE, TEARDOWN, OPTIONS, etc., as the name implies, it can be known as a dialogue and control function;

　　 During the RTSP conversation, SETUP can determine the port used by RTP/RTCP, PLAY/PAUSE/TEARDOWN can start or stop the sending of RTP, etc.;

　　 TCP, the full name is Transfer Control Protocol, and the Chinese name is Transmission Control Protocol. It works on the OSI transport layer and provides connection-oriented reliable transmission services.

　　 TCP's work is mainly to establish a connection, and then receive data from the application layer program and transmit. TCP uses virtual circuit connection to work. Before sending data, it needs to establish a connection between the sender and the receiver. After the data is sent, the sender will wait for the receiver to give a confirmatory response, otherwise the sender will think This data is lost, and resend this data.

RTP is not like http and ftp that 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 is played, it cannot be played repeatedly. , Unless you request data from the server again.

　　 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.

　　 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 a network application layer protocol similar to the http protocol.

Sequence number: indicates the position of the segment in the sequence of segments to be transmitted

Confirmation number: specifies the sequence number of the successfully received segment, the confirmation sequence number contains the next sequence number that the end sending the confirmation expects to receive

TCP offset: specifies the length of the segment header. The length of the section header depends on the option set in the section header option field

PSH: Indicates that the data will be sent to the receiving process as soon as possible

Window: Specify the command about the size of the next segment that the sender can transmit

Checksum: The checksum contains the TCP segment header and data part, used to verify the reliability of the segment header and data part

Emergency: indicates that the segment contains emergency information, and the emergency pointer is valid only when the U R G flag is set to 1.

Options: The recognized segment size, timestamp, the end of the option field are specified, and the boundary option of the option field is specified

TCP connection establishment: The TCP connection establishment process is also called TCP three-way handshake. First, the sender host initiates a synchronization (SYN) request to establish a connection to the receiver host; the receiver host replies with a synchronization/acknowledgement (SYN/ACK) response to the sender host after receiving this request; the sender host receives this After the packet is sent an acknowledgment (ACK) to the receiver host, at this time the TCP connection is successfully established;

TCP connection closure: After the sender host and the destination host establish a TCP connection and complete the data transmission, a data packet with the end flag set to 1 will be sent to close the TCP connection and release the buffer space occupied by the connection at the same time; TCP reset Setting: TCP allows the connection to be interrupted suddenly during transmission, which is called TCP reset;

TCP data sorting and confirmation: TCP is a reliable transmission protocol. It uses sequence numbers and confirmation numbers to track the data reception during transmission;

TCP retransmission: In the process of TCP transmission, if the receiver host does not receive an acknowledgement response to a data packet within the retransmission timeout period, the sender host considers the data packet lost and sends the data packet to the receiver again Side, this is called TCP retransmission;

TCP delay confirmation: TCP does not always confirm the data immediately after receiving it. It allows the host to send its own confirmation message to the other party while receiving the data.

TCP data protection (checksum): TCP is a reliable transmission protocol, which provides checksum calculation to realize the integrity of data during transmission.

　　 UDP protocol is the abbreviation of English UserDatagramProtocol, that is, user datagram protocol, which is mainly used to support network applications that need to transmit data between computers. Numerous client/server network applications including network video conferencing systems need to use the UDP protocol. The UDP protocol has been used for many years since its inception. Although its initial brilliance has been obscured by some similar protocols, even today, UDP is still a very practical and feasible network transport layer protocol.

　　 Like the well-known TCP (Transmission Control Protocol) protocol, the UDP protocol is located directly on top of the IP (Internet Protocol) protocol. According to the OSI (Open System Interconnection) reference model, UDP and TCP are both transport layer protocols.

　　 The main function of the UDP protocol is to compress network data traffic into the form of datagrams. A typical datagram is a transmission unit of binary data. The first 8 bytes of each datagram are used to contain header information, and the remaining bytes are used to contain specific transmission data.

　　 TCP is a point-to-point protocol, which means that each client needs to separate the client/server link, so data broadcasting to multiple clients cannot be realized at the network level. If a data stream must be transmitted to multiple clients at the same time, the server must transmit a copy of the data stream to each client. TCP can dynamically adjust the transmission speed according to the network bandwidth and the degree of congestion and resend the lost data packets. The reliability of data transmission is ensured, but server resources are expensive, and it is difficult to ensure the real-time performance of data stream transmission when the data stream is large.

UDP is an unreliable transmission protocol. At the sending end, the speed at which UDP transmits data is only limited by the speed at which the application generates data, the computer's capacity and transmission bandwidth; at the receiving end, UDP puts each message segment in a queue. The application reads a message segment from the queue each time; the UDP protocol does not need to maintain the connection state, and does not think that every data packet must reach the receiving end, so the network load is smaller than TCP, and the transmission speed is faster than TCP; The more congested the network, the more data packets are lost.

　　 The main difference between UDP and TCP protocol is how to achieve reliable transmission of information. The TCP protocol contains a special delivery guarantee mechanism. When the data receiver receives the information from the sender, it will automatically send a confirmation message to the sender; the sender will continue to transmit other information only after receiving the confirmation message. Otherwise, it will wait until the confirmation message is received.

　　 So TCP has more time to establish a connection than UDP. Compared with UDP, TCP has higher security and reliability. The size of TCP protocol transmission is not limited. Once the connection is established, both parties can transmit a large amount of data in a certain format, while UDP is an unreliable protocol with a size limit, which cannot exceed 64K each time.

　　 Compared with the TCP protocol, another difference of the UDP protocol is how to receive multiple datagrams that are unexpected. Unlike TCP, UDP does not guarantee the order of sending and receiving data.

RTP is above UDP. Although UDP is not as reliable as TCP and cannot guarantee the service quality of real-time services, RTCP needs to monitor data transmission and service quality in real time. However, because the transmission delay of UDP is lower than that of TCP, it can be very compatible with video and audio. Good match. Therefore, in practical applications, RTP/RTCP/UDP is used for audio/video media, and TCP is used for the transmission of data and control signaling.

The RTMP protocol is a protocol designed specifically for the efficient transmission of video, audio and data. It realizes real-time video and sound transmission by establishing a binary TCP connection or connecting an HTTP tunnel.

　　RTMP supports more media protocols than traditional media servers. It supports the dynamic transmission of multiple lines that may contain audio, video, and script data from the server to the client and from the client to the server. RTMP processes audio, video, and script data separately.

　　Sound and video data are buffered separately in the server. If the sound data reaches a certain limit in the sound buffer, all data in the buffer will be discarded, and the most recently arrived data will be allowed to start collecting in the buffer and sent to each client. Video data is processed in a similar way, the difference is that when a new key frame arrives, the data in the buffer is cleared. When discarding the old frame data, if it is found that the data of the client is wrong, the new and old frames are fitted.

　　RTMP gives different priority levels to data. In real-time conversation, sound is the most important, video is given low priority, and script data is given a priority between sound and video.

　　 The RTMP protocol can create multiple data streams, but each data stream can only have one direction. Using RTMP can build such a system, the client can interact with the RTMP server and the application server at the same time, so that the load on the server can be dispersed, although in this improved system structure, the performance requirements of the RTMP server are relatively high.

　　 HTTP protocol, the full name is HyperText Transfer Protocol, and the Chinese name is HyperText Transfer Protocol;

　　MMS protocol, the full name is Microsoft Media Server Protocol, and the Chinese name is Microsoft Media Server Protocol;

　　 HLS protocol, full name HTTP Live Streaming, is a streaming media transmission protocol based on HTTP implemented by Apple Inc.;

How far(long) the transmitter cover?

The transmission range depends on many factors. The true distance is based on the antenna installing height , antenna gain, using environment like building and other obstructions , sensitivity of the receiver, antenna of the receiver . Installing antenna more high and using in the countryside , the distance will much more far.

EXAMPLE 5W FM Transmitter use in the city and hometown:

I have a USA customer use 5W fm transmitter with GP antenna in his hometown ,and he test it with a car, it cover 10km(6.21mile).

I test the 5W fm transmitter with GP antenna in my hometown ,it cover about 2km(1.24mile).

I test the 5W fm transmitter with GP antenna in Guangzhou city ,it cover about only 300meter(984ft).

Below are the approximate range of different power FM Transmitters. ( The range is diameter )

0.1W ~ 5W FM Transmitter ：100M ~1KM

5W ~15W FM Ttransmitter : 1KM ~ 3KM

15W ~ 80W FM Transmitter : 3KM ~10KM

80W ~500W FM Transmitter : 10KM ~30KM

500W ~1000W FM Transmitter : 30KM ~ 50KM

1KW ~ 2KW FM Transmitter : 50KM ~100KM

2KW ~5KW FM Transmitter : 100KM ~150KM

5KW ~10KW FM Transmitter : 150KM ~200KM

How to contact us for the transmitter?

Call me +8618078869184 OR
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1.How far you want to cover in diameter ?
2.How tall of you tower ?
3.Where are you from ?
And we will give you more professional advice.

About Us

FMUSER.ORG is a system integration company focusing on RF wireless transmission / studio video audio equipment / streaming and data processing .We are providing everything from advice and consultancy through rack integration to installation, commissioning and training.

We offer FM Transmitter, Analog TV Transmitter, Digital TV transmitter, VHF UHF Transmitter, Antennas, Coaxial Cable Connectors, STL, On Air Processing, Broadcast Products for the Studio, RF Signal Monitoring, RDS Encoders, Audio Processors and Remote Site Control Units, IPTV Products, Video / Audio Encoder / Decoder, designed to meet the needs of both large international broadcast networks and small private stations alike.

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We have customers of public and commercial broadcasters, telecom operators and regulation authorities , and we also offer solution and products to many hundreds of smaller, local and community broadcasters .

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