Selection of transmission layer protocols for digital video Networks

Abstract: With the increasingly mature video compression technology, digital video monitoring products have gradually become the mainstream of the market. People Urgently hope that the network technology can be successfully applied to the field of digital video surveillance, so that people can achieve remote monitoring through the network. In this context, the study of "network-based digital video transmission" is becoming increasingly important. This article focuses on the real-time and quality of network video transmission, starting from "solving the transport layer protocol problem", through the comparison of TCP protocol and RTP protocol, the design concept of "RTP real-time video transmission based on UDP protocol" is proposed, which ensures the real-time and service quality of digital video transmission.

 

Key words: Digital Video; Network; Transport Layer Protocol

 

1 Introduction

 

Digital video transmission is a communication method that people use vision to obtain information. Compared with other information transmission methods, digital video transmission is accurate, intuitive, and efficient. Due to the large amount of information transmitted by digital videos and limited bandwidth, the compression and encoding of videos, selection of transmission channels and network protocols, and IP Multicast technology (IP Multicast) software Implementation Based on the Windows operating platform has become a key technology in network-based digital video transmission applications. Among them, the choice of transmission channels and network protocols is crucial, it will directly affect the real-time performance of digital video transmission and the quality of the video image received by the client after transmission through the network.

 

 

 

Figure 1 shows the basic network-based digital video transmission model. This article will analyze the selection of network transmission protocols for network-based digital video transmission applications.

 

 

2. The selection of network transmission protocols is concentrated at the transport layer.

 

We know that the OSI (Open System Interconnection) network reference model developed by ISO divides the network into seven layers, which are listed in the following order: physical Layer, data link layer, network layer, transmission layer, Session Layer, presentation layer, and application layer. The functions of each layer are independent of each other. The functions implemented by each layer are transparent to the previous layer, and each layer only cares about the services provided by the next layer.

 

Physical Layer tasks transmit bit streams transparently. data link layer tasks transmit frame-based data without errors on the lines between two adjacent nodes through various protocols, their functions have been solidified by the transmission media and network card and cannot be changed through programming. Therefore, the two layers do not need to be considered during programming. The task at the network layer is to select an appropriate route, to package and transfer data, Netware uses the IPX protocol at this layer, while UNIX and Windows use the IP protocol. The task at the transport layer is to best utilize network resources based on the characteristics of subnets, in addition, it establishes a transmission connection for the two hosts in a reliable and economical way to transmit packets transparently. At this layer, Netware uses the SPX protocol, while Windows uses the TCP/UDP protocol, UNIX uses the TCP protocol. The Session Layer task is used to determine the transmission mode of information between the interconnected hosts, namely, full duplex, half duplex, and ticket. Generally, this parameter is not considered in the LAN; the presentation layer task is to format the transmission data and convert the code. It is generally used for communication between different machines, so there is no such problem between the microcomputer, so this layer does not need to be considered; the application layer Task is to determine the nature of communication between processes to meet user needs. It is a programming task, but no protocol selection is required. From this point of view, in order to ensure the real-time and image quality of digital video transmission based on the network, the choice of transport layer protocol is the key to the entire design and implementation.

 

The Internet uses two transmission protocols on the IP layer: one is the transmission control protocol TCP (Transmission Control Protocol), which is a connection-oriented network protocol; another is the user data Datagram Protocol (UDP), which is a connectionless network protocol. Figure 2 shows the concept hierarchy of networks:

 

 

 

Before analyzing and comparing the transport layer protocols, we need to understand the concepts of "connection-oriented" and "No connection. Connection-oriented service is the abstraction of the telephone system service model, that is, every complete data transmission must go through the process of establishing, using, and terminating a connection. During data transmission, each data group uses a connection ID instead of a destination address ). Essentially, a connection is a pipe, and the data sent and received are not only in the same order, but also in the same content. TCP provides connection-oriented virtual circuit services.

 

The connectionless service is the abstraction of the postal system service. Each group carries a complete Destination Address and each group is transmitted independently in the system. The connectionless service does not guarantee the order of groups. It does not recover and retransmit group errors, and does not guarantee the transmission reliability. UDP provides the connectionless datagram service.

 

3. TCP is not suitable for transmitting audio and video data in real time.

 

The IP network has been widely used in various scenarios. Among them, the TCP/IP protocol is the industrial standard for the interconnection and communication of Operating Systems in heterogeneous networks. The system is built on TCP/IP to broaden its application scope. However, the pure TCP/IP protocol has been difficult to adapt to the requirements of video and audio communication, especially continuous media streams (such as video streams) communication. TCP is a connection-oriented protocol used to provide virtual circuit services for ordered and reliable data transmission on various networks. Its retransmission Mechanism and Congestion Control Mechanism are not suitable for real-time video and audio transmission.

 

The TCP/IP protocol was initially designed to provide non-real-time data services. The IP protocol is responsible for data transmission between hosts without error detection or error correction. Therefore, data is often lost or out of order. To ensure reliable data transmission, people use the TCP protocol for IP data transmission to improve the error detection and Error Correction capabilities at the receiving end. When packet loss or errors are detected, the sender is required to resend the packet, which inevitably leads to transmission latency and network bandwidth consumption. Therefore, traditional TCP/IP protocols provide poor ability to transmit real-time audio and video data. Of course, TCP is also an option when transmitting the video and audio data used for playback. If there is enough buffer and enough network bandwidth, it is also possible to transmit close to real-time video and audio over TCP. However, if the packet loss rate is high and the network condition is poor, it is almost impossible to use the TCP protocol for video or audio communication.

 

TCP and other reliable transport layer protocols such as xtp are not suitable for real-time video/audio transmission, mainly due to the following reasons:

 

3.1 TCP retransmission mechanism

 

We know that in TCP/IP, when the sender finds data loss, it will require re-transmission of the lost data packet. However, this requires one or more cycles (according to the fast retransmission mechanism of TCP/IP, this requires three additional frame latencies ), this retransmission is almost disastrous for video and audio data communication with high real-time requirements, because the receiver has to wait for the arrival of retransmission data, this results in latency and breakpoint (audio discontinuous or video solidification, etc ).

 

3.2 TCP congestion control mechanism

 

TCP's congestion control mechanism reduces its congestion window when detecting packet loss. On the other hand, the number (Bit Rate) produced by audio and video in a specific encoding method cannot be changed suddenly. Correct congestion control should be to change the encoding method of audio and video information, and adjust the frame rate or image width of video information.