The performance parameters of jitter are mainly the following three types: This article refers to the address: http:// (1) Input jitter tolerance period The maximum jitter range of the input signal allowed by the system. (2) The signal jitter characteristic of the output of the jitter system. (3) Jitter transfer characteristic The distribution characteristic (gain) of the jitter value of the input signal of the system and the jitter value of the output signal at different test frequencies. 6, reliability and longevity People are accustomed to using the technical performance indicators of products as a measure of the quality of products. But this is only one aspect, and it does not reflect the overall picture of product quality. Only products that have good technical performance, are durable, and fully reliable can be considered as truly good quality products. The latter is usually measured by the reliability and longevity of the product. For a digital system, product reliability and longevity include the reliability and longevity of all products on the transmission path between the two terminal interfaces of the system. Through the actual investigation of the use site, counting the number of failures, recording the interval between each failure, and the maintenance time of each failure, the average time between failures and average maintenance time of the system can be obtained. Reliability is the ability of a product to perform a defined function under specified conditions and time. Common failure rate (λ) or mean time between failures and average repair time. The failure rate (λ) indicates the probability of failure per unit time under the condition that the product works to the time t. The commonly used 10-9/h is used as the reference unit and is called a fit. Mean Time to Life (MTTF) is the average working time before a device or system fails. Mean Time Between Failure (MTBF) refers to the interval between two adjacent failures. Mean Time To Repair (MTTR) is the time required to troubleshoot. Seven, digital signal compression As the amount of information transmitted increases, frequency resources become more and more tense. For example, high definition television, video conferencing, e-books, multimedia, and digital broadcasting all occupy a relatively wide frequency band. Assuming that one channel television signal is transmitted, the highest frequency of the luminance signal is set to 6 MHz, and the maximum frequency of the color difference signals (RY and BY) is set to 1.5 MHz. According to the sampling theorem, the sampling frequencies of the luminance signal and the color difference signal should be 12 MHz and 3 MHz, respectively. If the number of quantization layers is 256, an 8-bit binary code is required to transmit the quantized value of each sample. Thus, the transfer rate should be (12 + 3) × 8 = 120 Mb / s. Such a high transmission rate occupies a very wide frequency band, which is about 100 MHz or more. Therefore, compression bandwidth is an important issue in digital television. If text, sound, graphics, and images are added, the amount of information data is very large, and such a large amount of data is processed in real time, and the transmission rate and the storage capacity are hard to reach. Therefore, the amount of data must be compressed efficiently, in real time, especially for video images. Each pixel of a television picture is related to each other, and there is correlation between adjacent pixels, between adjacent lines, and between adjacent frames in the same row, which indicates that there is redundancy in the information, and the redundancy is removed or reduced. Redundancy compresses the transmission code rate and also compresses the frequency band. Image redundancy is divided into spatial redundancy and time redundancy. Spatial redundancy refers to the regular image in 1 frame or similar content in the image. These pixels have strong correlation, so intra-frame compression can be performed, and only the part with no correlation is transformed and encoded; time redundancy refers to The content similar to the image between frames, the statistics show that the amount of pixel change between frames is generally only about 10%, so the time domain correlation between the front and back frames of the video image can be used for inter-frame compression, that is, from the past few frames. The content predicts the next frame of image, and only predicts and encodes data that is different from the previous frames. In addition, the human eye is less sensitive to clutter with too high a frequency component, and is sensitive to the brightness of the image and weak to the color resolution. The human eye has low sensitivity to the edge of the image, so the sensitivity can be lower. Compress. You can also make unwanted or unwanted selections for certain complex images and fast-changing images to reduce the amount of data. Spatial redundancy compression coding needs to first divide the image into sub-regions. The size of each sub-region can be 1×4, 1×8, 1×16 pixels (one-dimensional case), or 4×4, 8 ×8, 16 × 16 pixels (two-dimensional case), typical transform coding is Discrete Cosine Transform (DCT). At present, there are three major image compression coding standards: JPEG (Joint Photographics Expert Group) standard, MPEG (Motion Picture Expert Group) standard and H.261, of which MPEG standard is the most commonly used standard. . 1, JPEG standard The JPEG standard is mainly for still images and is also applied to moving images. Its compression is relatively low. For a medium-complex color image, the relationship between compression ratio and image quality is shown in Table 3. 2, MPEG standard The MPEG standard mainly includes MPEG-1 and MPEG-2 standards, and MPEG is an abbreviation of Motion Picture Expert Group. MPEG-1 is an encoding with a code rate of 1.5 Mb/s; MPEG-2 is an encoding with a code rate of 10 Mb/s. With the development of technology, MPEG-4 has been developed and studied, which is a low bit rate (<64 kb/s) video and audio coding. The most striking feature of the MPEG standard is MPEG-2, which is a universal encoding standard for moving pictures and accompanying sounds, suitable for various applications such as storage media, distribution transmission and communication. There are 9 parts in the MPEG-2 standard. Part 1: System; Part 2: Video; Part 3: Audio; Part 4; Compliance testing; Part 5: Software; Part 6: Instructions and controls for digital storage media; Post-compatible audio; Part 8: 10-bit video; Part 9: Real-time interface. The second part of the video is the core of the standard, which uses a combination of interframe and intraframe compression methods. 1) Reduce time domain redundancy. MPEG divides moving pictures into I frames (intra pictures), P frames (predicted pictures), and B frames (interpolated pictures). The I frame is only used for intraframe compression coding. After moderate compression, it can be used as the starting point of random access. Therefore, the I frame image must be transmitted. The P frame is only used for forward interframe prediction, and the prediction error is conditionally transmitted. It is higher than I frame; B frame is referenced to I frame and P frame for bidirectional inter-frame prediction interpolation, and its compression degree is high. 2) Reduce airspace redundancy. MPEG is based on DCT compression coding, which reduces the spatial redundancy. It is basically the same as the JPEG standard. The whole process is divided into three steps: DCT orthogonal transform based on macroblock to calculate transform coefficients; scalar quantization of transform coefficients , re-combining data in the "Z"-shaped scan order; encoding the data to further compress the data. Unlike JPEG, video information in MPEG includes not only still picture pictures (intra pictures) but also motion information (inter prediction pictures, interpolation pictures). The MPEG-1 standard has an average compression ratio of 50:1 and an image resolution of 352 × 240 pixels (30 frames / s) or 352 × 288 pixels (25 frames / s), mainly used to compress still images. MPEG-2 has an image resolution of 720 x 480. Stereo is compatible with MPEG-1 standard audio and is mainly used to compress moving images. 3. H.261 standard The H.261 standard is used for image compression coding of video telephony and video conferencing, with an integer multiple of 64 kb/s as the transmission rate, and the image resolution and frame change rate can be adjusted. This standard is not applicable to broadcast television. Digital compression images and non-compressed images have some disadvantages, such as: 1) Some digital image information will be lost when the digital compression image changes at high speed. In addition, if the digital compressed image is misinterpreted during transmission, the quality of the image restored by the receiving end is greatly affected, that is, there is a problem of error spreading unique to compression encoding. 2) Since the compression coding cancels the line and field blanking signals, it makes it impossible to use the line and field blanking period to transmit information. Eight, digital video broadcasting standards Due to the high cost of high definition television (HDTV), countries around the world have adopted the quality of broadcast TV between analog TV and HDTV in order to unify digital video and its related interactive services worldwide before implementing HDTV. The standard for digital video broadcasting (DVB, Digital Video Broadcast). These standards were developed by the Digital Video and Audio Commission (DAVIC), a cross-border, cross-industry international organization with more than 300 members in more than 30 countries. At present, various digital video broadcasting systems based on the DVB standard have begun to operate in Europe, America, Asia, Africa and Australia, and will enter a more extensive implementation and expansion stage. Various media such as satellite, microwave, cable TV and terrestrial broadcasting comply with digital video broadcasting standards, which are important for equipment development, production, cable operators and users. For example, a standard digital user terminal will lead to competition between equipment manufacturers and suppliers, bringing price and quality benefits to users. Nowadays, DVB products produced by companies such as Philips, NDS, ScintificAtlanta and Thomson have been available on the market. The core content of the DVB standard is the adoption of the MPEG standard in source coding, video and audio compression and multiplexing. In the channel coding and modulation part, a series of standards are developed. RS (Reed Solomon, Reed-Solomon) forward error correction code and convolutional error correction code are used to select the corresponding modulation mode and channel according to different media. coding. These standards include: DVB-S: For satellite TV broadcasting, using the 11/12 GHz band. DVB-C: For cable TV broadcasting. DVB-PI: For DVB and cable TV front ends and SMATV front end interfaces. DVB-T: For terrestrial open circuit television broadcasting. DVB-TXT: for digital TV graphic systems. DVB-SI: for service information systems. DVB-MC: For multi-point microwave distribution systems (MMDS), using bands below 10 GHz. DVB-IRDI: Integrated Receiver Decoder (IRD) interface. There are other standards such as DVB Test (DVB-M), Scrambling and Conditional Access Interface (DVB-CA). The standards that DVB is improving include Interactive Television System Standard (DVB-I), Interactive System Network Protocol (DVB-NIP), Interactive System Cable TV (CATV) and Public Switched Telephone Network (PSTN) and Integrated Services Digital Network (ISDN). ) Channel standards, DVB and SDH and ATM interface standards. The DVB standard is not only applicable to today's digital TVs, but also to high definition television (HDTV) in the future, with a bit rate of 60 Mb/s. Low-definition TVs only need 4Mb/s, and standard definition TVs are 15Mb/s. As shown in Figure 6, the DVB system first uses MPEG-2 data compression to achieve band compression of TV images. TV audio is often MPEG-1 data compression, and then low error of TV image signals according to DVB channel coding standard. Rate transfer. Figure 6 Schematic diagram of digital TV broadcasting system MPEG-2 coding belongs to source coding, which mainly specifies compression and multiplexing of digital signals. For example, inter and intra motion compensation (without time redundancy), discrete cosine transform (organized information content), variable length run length coding (with information redundancy removed). In the multiplexing aspect, the multiplexing mode, synchronization mode, and data packet composition (188 bytes) of the video and audio data streams are specified. For the 500-line normal horizontal definition TV image, if the general digital coding is used, the sampling frequency and transmission code rate are as shown in Table 4. If MPEG-2 compression coding is used, the compression rate is only 2-2 when the compression ratio is 12-62. 10Mb/s. 1. DVB-S satellite digital TV The source of satellite digital television broadcasting is shown in Figure 7. Multi-channel audio A, video V, or data signal is first passed through MPEG-2 digital encoding compressor, compressing rate above 200Mb/s to below 6Mb/s, and multi-channel compressed MPEG-2 data stream is sent to digital multi-channel The multiplexer mixes to obtain a MPEG-2 digital stream that is much higher than 6Mb/s and is finally sent to the channel coding portion. If the television program needs to be scrambled, the TV scrambled signal can be sent to the multiplexer. The main technical performance of the MPEG-2 digital encoding compressor in the figure is as follows: Input signal PAL (or NTSC) composite analog signal, video, audio analog signal; Resolution CCIR601 to SIF; Aspect ratio 4:3, 16:9; Bit rate 1.5-15 Mb/s, variable; Output E-1, E-3, DS-3, OC-3, ATM optional, 2 serial output ports, 1 parallel output port. The main technical performance of the digital multiplexer is as follows: Input signal 4 channels, RS232 or RS422 data stream (600-19200b/s), Ethernet code stream (2.1Mb/s), TV scrambled signal; The output can be connected to a QPSK or QAM modulator and can be sent to the multiplexer again. There are 2 MPEG-2 stream output ports, E-3, DS-3, OC-3, ATM are optional. Figure 7 DVB-S source coding block diagram The channel coding of DVB-S satellite digital television mainly uses convolutional code, RS code and other coding methods. The modulation mode is QPSK modulation, output 70MHz intermediate frequency, and finally upconverted to satellite channel, as shown in Figure 8(a). Figure 8(b) is a block diagram of the channel decoding at the receiving end. Contrary to Figure 8(a), the output is an MPEG-2 digital stream. The Integrated Receiver Decoder (IRD) for receiving satellite digital television consists of an MPEG-2 decoder and a channel decoder, which must comply with both the DVB-S standard and the MPEG-2 standard. Figure 8 DVB-S system block diagram The main performance of the QPSK modulator is shown in Table 5. 2, DVB-C cable digital TV Since cable television signals are transmitted in coaxial cable and optical fiber, the external interference is small, and DVB-C specifies the use of QAM modulation. The main contents of channel coding and modulation are: 1) spectrum diffusion. A pseudo-random sequence is added to the data stream, and its spectrum is like the spectrum of white noise. 2) Adopt RS coding. Add 16 guard bytes to the 188 bytes of the packet for forward error correction. 3) Bit interleaving coding. Depth 12 bytes to correct burst errors. 4) QAM modulation. According to the transmission environment, 16-QAM, 3-2QAM, 64-QAM, and 258-QAM can be used. Systems with long transmission distance and high noise use low-rate 16-QAM and 3-2QAM, and vice versa with high-speed 64-QAM and 258-QAM. For example, 64-QAM has a transmission rate of 38.5 Mb/s in an 8 MHz bandwidth. 5) Square root Nyquist filter. The roll-off rate is 15% to balance channel utilization and nonlinear distortion. The main technical performance of the M-QAM digital modulator is shown in Table 6, M=16, 32, 64, 128, 256. If the signal source is the AV output of the recorder, it is converted into a data stream by the MPEG-2 encoding compressor; if it is a satellite analog TV, the AV signal is output through the tuner and the satellite receiver, and converted by the MPEG-2 encoding compressor. Into the data stream, the multiplexed data stream is mixed into the MPEG-2 code stream by the digital multiplexer, and the QAM modulation is used, and the output intermediate frequency is 36 MHz or 44 MHz, and then up-converted to the cable channel. If it is a satellite digital TV, it is sent to the QPSK-QAM modulation converter after being down-converted by the tuner. (The main technical performance of the QPSK-QAM modulation converter is shown in Table 7), and the QPSK is converted into a QAM modulation signal, and then passed through the up-converter. Upconvert to cable TV channels. Finally, the multi-channel cable channels are mixed into the cable network or optical transmitter, and can also be sent to the broadband microwave transmitter. 3, DVB-T open circuit digital TV In the open analog TV band (UHF, VHF), due to the multiple reflections of buildings and mountains and various noise interferences, the quality of open-circuit television is not high, and the frequency resources are seriously insufficient. Therefore, it is necessary to improve to open-circuit digital TV. The open digital TV system consists of MPEG-2 encoding compressor, forward error correction (FEC), quadrature encoding frequency division multiplexer (COFDM), digital modulator, and upconverter. The final output frequency band is still UHF and VHF. Channel, as shown in Figure 9. In addition to MPEG-2 encoding compressors, forward error correction and satellite digital television and cable digital television, orthogonal coded frequency division multiplexers are features of open digital television. These features are: (1) Multi-carrier mode in the frequency domain, with 2000 or 8000 carriers. (2) The digital baseband signal is divided into segments in the time domain to modulate a plurality of carriers. (3) All transmitting stations that constitute a certain transmission network of DVB-T are locked to a precise frequency by the Global Positioning System (GPS), so that all transmitters use the same frequency and transmit the same bit information at the same time. Figure 9 Block diagram of open circuit digital television broadcasting system The open circuit digital TV can adopt QPSK modulation or QAM modulation, and the output is up-converted and sent to the transmitting antenna to transmit to the space. Nine, the main network standard Information transmission includes transmission, switching and access. The CATV is the broadband access part of future information transmission, and may also become a complete information transmission network. Broadband access networks are ideal for delivering analog and digital services such as telephony, cable television, data (conference TV, HDTV, TV shopping) and more. In order to connect with other information transmission networks, CATV must also comply with the unified network standards. Considering the future access of cable television networks to the Internet or WAN, several major network standards and specifications are now briefly introduced, including transmission and reception. Digital TV is also inseparable from these major network standards. A network that connects multiple computers and communicates with each other in a local area (such as a company, a building, or a large hotel) to share hardware resources and data information is called a local area network (LAN). Currently, the most LAN in the world is Ethernet (Ethernet, a type of local area network). The Ethernet standard is a plurality of standards in the IEEE802.3 series, including 10BASE, 100BASE, 1000BASE, etc., which successfully solve the computer interface. , connection media and information transfer issues. Due to the loss of signal on the line, various LAN technologies limit the coverage of the LAN, such as 10BASE5 maximum transmission distance of 500m; 10BROAD36 is suitable for cable TV transmission, the maximum distance is 3600m, and the transmission rate can reach up to 10Mb/s. In order to achieve long-distance connection of computers, the Wide Area Networks (WAN) standard has been developed, including X..25, Frame Relay, ATM, IDSN, SDH and other standards (or protocols), mainly using high-speed modems and digital subscriber lines. (DSL) to transmit signals and use a dedicated computer to process and transmit signals to local multimedia computers. 1. Internet (Internet) Protocol In order to interconnect incompatible WANs and LANs into the Internet, uniform international standards are needed to coordinate the operation of software and hardware that conform to their respective network standards (or protocols). The two main standards are the Transmission Control Protocol (TCP) and the Internet Protocol (IP), collectively referred to as Transmission Control/Internet Protocol (TCP/IP). Compared with other networks, IP networks have the following characteristics: they have a high degree of independence for specific protocols of subnets, and there is strong independence between subnets. Subnets are highly independent of user-specific protocols. Therefore, the IP subnet can be various networks such as Ethernet, Frame Relay, and ATM. IP is a connectionless operation protocol, and TCP is a connection-oriented protocol. Connectionless operation means that no logical connection is established between the user and the network before data transmission, and each data unit is transmitted as a separate unit. Thus, data units can take different paths to avoid faulty nodes and congested nodes in the network. Connection-oriented operation means that before data transmission, a logical connection is first established between the user and the network, and then the data unit is transmitted through the logical connection, and the data units maintain a certain relationship. Due to the long header information of IP data units, the overhead of "bits" is large, and the sequential control and acknowledgment functions are not supported (or rarely supported), user data may be lost, duplicated, or even reversed. The transmission control protocol TCP can solve these problems. It can perform tasks such as reliability control, flow control, and sequence control well. Therefore, TCP and IP are so closely connected that TCP and IP are closely connected to become TCP/. IP protocol. The familiar e-mail (E-mail), file transfer (FTP), and remote login (Telnet) protocols are also compatible with TCP/IP and are TCP/IP application layer protocols. 2. Quasi-synchronous digital system At present, there are many places in the telecommunication network that use the Plesiochronous Digital Hierarchy (PDH). The PDH has two rates of 1.544Mb/s and 2.048Mb/s. It is a set of digital transmission code streams that are grouped into 64Kb/s telephone channels. . It is designed primarily for voice services and also for video transmission. The shortcomings of the pseudo-synchronous digital system (PDH) are as follows: (1) Only two base signals of 1.544Mb/s and 2.048Mb/s are synchronously multiplexed, and the other high-speed levels are asynchronously multiplexed. In order to fetch or insert a data stream, multiplexing and demultiplexing must be performed over a wide range to adjust the code rate. (2) It is mainly a point-to-point connection. The multiplexed structure makes it difficult to change the line transmission after other nodes fail, and it is difficult to measure network performance and manage remote network elements. (3) There is not enough bandwidth. (4) There are several independent regional standards in the world, and it is difficult to achieve international interoperability. With the development of the information society, the quasi-synchronous digital system can no longer meet the needs. Therefore, the information transmission nowadays uses a large number of synchronous digital systems. 3. Synchronous digital system Synchronous Digital Hierarchy (SDH) consists of several basic network elements. It is a multi-purpose information network that integrates multiplexing, transmission, switching, and network management. Its core is synchronous multiplexing, information structure, and division. Plug-in and cross-connect, unified optical interface, with a unified network node interface, such as 64Kb / s telephone nodes, broadband switching nodes. Synchronous digital system is not only suitable for optical cable, but also for microwave and satellite transmission; not only for long-distance network, but also for relay and access network; not only fully compatible with PDH, but also accessible to asynchronous transfer mode (ATM, Asynchronous Transfer) Mode), Broadband Integrated Services Digital Network (B-ISDN), Synchronous Optical Network (SONET) transmitted information. The information structure of SDH is divided into several standard levels, called synchronous transmission module STM-1, with a transmission rate of 155.20 Mb/s; four STM-1 synchronous multiplexing constitutes STM-4, and the transmission rate is 4×155.20 Mb/s=622.080. Mb/s; 4 STM-4 (or 16 STM-1) synchronous multiplexing constitutes STM-16, the transmission rate is 4 × 622.080 Mb / s = 2488.320 Mb / s; and so on, to get STM-64 .... SDH has a powerful network management system for performance monitoring, network maintenance, and interoperability of different company devices. The digital audio system SDH video and audio encoder can receive the baseband signal, while the intermediate frequency image, the scrambled image, and the image signal multiplexed and modulated by MPEG-2 compression cannot directly input the video and audio coding of SDH. Device. Therefore, devices such as on-board converters and scramblers should be added to the SDH network distribution point. Although the SDH equipment produced by each company has the same interface, the encoding method of the encoder and the decoder is dedicated to each company and cannot be compatible, so it is not easy to mix and use in one system. From the point of view of use, there are the following differences between SDH and PDH: (1) The rate is different. PDH has 2Mbps, 18Mbps, 34Mbps and 140Mbps (equivalent to 1920 voice channels); SDH has 155Mbps, 1440Mbps, 2488Mbps and 9953Mbps. (2) Different multiplexing methods. PDH multiplexing is one-level first-level multiplexing or de-multiplexing, such as multiplexing two 2Mbps signals into one 8Mbps, four 8Mbps multiplexing to one 34Mbps, and four 34Mbps multiplexing to one 140Mbps; On the contrary, demultiplexing proceeds step by step. SDH multiplexing is done once, multiplexed to 155 Mbps from 2 Mbps, and vice versa, separating 2 Mbps from 155 Mbps. (3) The optical interface standards are different. P DH equipment optical interface standards are not uniform for each company, so each company's equipment must undergo photoelectric and electro-optical conversion, thereby adding a large number of equipment. SDH unifies the optical interface standard, and optical signals in different regions do not need to be photoelectrically converted. (4) The PDH does not have enough channels to centrally manage and schedule all networks; SDH has enough channels to store network management information. SDH enables high-speed and large-capacity digital transmission, long-distance video telephony, interactive TV, and remote medical diagnosis. However, SDH has some disadvantages, such as high frequency band utilization, poor PDH performance, poor jitter performance, and excessive software rights. 4, asynchronous transfer mode Asynchronous Transfer Mode (ATM) is a network standard for implementing broadband digital integrated services. It uses a high-speed virtual circuit-oriented cell switching method, in which each cell is a fixed 53-bit length, and the first 5 bits are used to describe the destination address. The other 48 bits represent information. This method can effectively reduce the delay and irregular delay of the information transfer process, so the ATM asynchronous transfer mode is suitable for various time-sensitive information transmission, such as voice, TV video and multimedia information. Its transmission speed has a large range (ATM's adjustable bandwidth is 64Kbps - 622Mbps), and it will reach Gigabit per minute in the future. The bandwidth can be dynamically allocated according to different needs. In order to adapt to different required transmission speeds, a hierarchical transmission mode is adopted for video, audio, and data, which can be applied to a local area network or a wide area network. In addition, ATMs can be extended directly to desktop computers to allow applications to take advantage of the many features of ATM. ATM has both a circuit switching system and a packet switching system. Although circuit-switched systems can maintain a certain bandwidth and time delay, frequency resources cannot be shared. In the initial telephone exchange system, each time a call is made, a special line must be established between the Talker and the Receiver. At the same time, there can only be one pair of users, even if the line is idle, other users cannot share. Subsequent time division multiplexing techniques divide each user into a specific time slot to transmit digital voice, but this time slot also has strict special features, that is, other users cannot share at the same time. Therefore, circuit switching systems waste a lot of frequency resources. With the rapid development of computer technology, another voice digital transmission technology has begun to emerge, which is packet switching technology. The information transmitted by the user is cut into short pieces, each of which contains address information and flows on the network in the form of a packet. At present, many companies have developed many ATM products, such as: 25M/100M network cards, ATM hubs, LAN analog programs, ATM WAN switches, and so on. These products will lay a solid foundation for the realization of film and television on demand, distance education and multimedia services on a large scale. Due to historical reasons, China's telecommunications sector operates equipment with many different protocols or standards on the WAN, such as devices that comply with DDN (Digital Data Network), X.25, ATM, Frame Relay, ISDN, ChinaNet, 169, etc. . In order to balance these devices, telecom companies have to develop network technologies that support multi-protocol and multi-service. This is the label-switching (MPLS standard) technology based on ATM. It supports both IP and multi-protocol, multi-service routing switches. They are building FTTSA (Fiber to the Service Area), and FTTH (Fiber to the Home) is the future development goal. A level 1 ring network is formed between the provincial capitals, and a level 2 ring network is formed between the cities and towns, and a level 3 network is formed between the county and the county. In the service area is a symmetric high bit rate digital subscriber line (HDSL) with an uplink and downlink transmission rate of about 2 Mb/s, which is transmitted in both directions for local area network. Or use asymmetric digital subscriber line (ADSL), downlink transmission rate of 2-8Mb / s, two-way transmission, uplink transmission rate of 160-640Kbps. China's cable TV network (including digital microwave) uses SDH to realize the province's networking. The main equipments include video servers, cable modems, routers, digital office machines (HDTs) and user interface units for on-demand broadcasting. Each cable TV station can use 10/100/1000 Mbps Ethernet and G-bit routing switches to build a broadband IP metropolitan area network, run IP on the SDH network, and then run IP over the optical cable network to achieve IP-based national networking. Bandwidth can also be extended if dense wavelength division multiplexing (DWDM) is used. Since most cable TV stations in China have built fiber-optic cable hybrid networks, there are redundant optical fibers in the trunk line of the cable, which are generally more than 2 cores (the spare fiber for network design, so these redundant fibers can be fully used to build a broadband IP network. 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