Basic principles and related technologies of the h

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VoIP basic principle and related technology

voice communication through Internet is a very complex system engineering, which has a wide range of applications, so there are many technologies involved. The most fundamental technology is VoIP (voice over IP) technology. It can be said that Internet voice communication is one of the most typical and promising application fields of VoIP technology. Therefore, before discussing voice communication with Internet, it is necessary to first analyze the basic principles of VoIP and the related technical problems in VoIP

first, the basic transmission process of VoIP

traditionally, voice is transmitted by circuit switching, and the required transmission broadband is 64kbit/s. The so-called VoIP uses the IP packet switching network as the transmission platform to compress and package the analog voice signal, so that it can be transmitted by using the connectionless u tensile strength testing machine: sealant tensile test report (Figure) DP protocol

in order to transmit voice signals on an IP network, several elements and functions are required. The simplest form of network is composed of two or more devices with VoIP function, which are connected through an IP network. The basic structure of VoIP model is shown in Figure 1. From the figure, we can find how VoIP devices convert voice signals into IP data streams, and forward these data streams to IP destinations, which convert them back to voice signals. The network of the voice of the two must support IP transmission, and can be any combination of IP router and network link. Therefore, the VoIP transmission process can be simply divided into the following stages

1. Voice data conversion

but in the laboratory? Is it feasible to use an expandable automatic system for material testing? This means that the

voice signal is an analog waveform, and voice is transmitted through IP. Whether it is real-time application service or non real-time application service, daomaoan must first convert the analog data of the voice signal, that is, quantize the analog voice signal by 8 or 6 bits, and then send it to the buffer storage area. The size of the buffer can be selected according to the requirements of delay and coding. Many low bit rate encoders encode in frames. The typical frame length is 10~30ms. Considering the cost of transmission, interlanguage packets usually consist of 60, 120 or 240ms of voice data. Digitalization can be realized by using various speech coding schemes. The currently adopted speech coding standard mainly includes ITU-T G.711. The source and destination speech coders must implement the same algorithm, so that the destination speech device can restore the analog speech signal

2. Original data to IP conversion

once the voice signal is digitally encoded, the next step is to compress and encode the voice packet with a specific frame length. Most encoders have a specific frame length. If an encoder uses 15ms frames, the 60ms packets from the first are divided into 4 frames and encoded in sequence. Each frame is composed of 120 voice samples (the sampling rate is 8kHz). After encoding, four compressed frames are synthesized into a compressed speech packet and sent to the network processor. The network processor adds packet headers, timestamps and other information to the voice and transmits it to the other end point through the network. The voice network simply establishes a physical connection (a line) between the communication endpoints and transmits encoded signals between the endpoints. Unlike circuit switching network, IP network does not form a connection. It requires that data be placed in variable length datagrams or packets, and then each datagram is attached with addressing and control information, which is sent through the network and forwarded to the destination station by station

3. Transmission

in this channel, all networks are regarded as one to receive voice packets from the input end, and then transmit them to the output end of the network within a certain time (T). T can change in a full range, reflecting the jitter in network transmission. The same node in the network checks the addressing information attached to each IP data, and uses this information to forward the datagram to the next station on the ground path that should be cooled quickly. A network link can be any topology or access method that supports IP data flow

4. IP packet data conversion

the destination VoIP device receives the IP data and starts processing. The network stage provides a variable length buffer to adjust the jitter caused by the network. The buffer can accommodate many voice packets, and the user can choose the size of the buffer. A small buffer produces a small delay, but cannot adjust a large jitter. Secondly, the decoder decompresses the encoded voice packets to generate new voice packets. This module can also operate by frame, which is completely the same as the length of the decoder. These implants are easy to cause inflammation. If the frame length is 15ms, 60ms voice packets are divided into 4 frames, and then they are decoded and restored into 60ms voice data stream and sent to the decoding buffer. In the process of datagram processing, the addressing and control information is removed, the original original data is retained, and then the original data is provided to the decoder

5. Digital voice is converted to analog voice

the playback driver takes out the voice samples (480) in the buffer and sends them to the sound card, which is broadcast at a predetermined frequency (such as 8kHz) through the speaker. In short, the transmission of voice signals on the IP network needs to go through the process of conversion from analog signals to digital signals, encapsulation of digital voice into IP packets, transmission of IP packets through the network, unpacking of IP packets, and restoration of digital voice to analog signals. The whole process is shown in Figure 2

second, the driving force for the development of VoIP

due to many developments and technological breakthroughs in related hardware, software, protocols and standards, the widespread use of VoIP will soon become a reality. The technological progress and development in these fields have contributed to the creation of a more effective, functional and interoperable VoIP network. The table briefly lists the main developments in these areas. It can be seen from the table that the technical factors that promote the rapid development and even wide application of VoIP can be summarized as follows

1. Digital signal processor

advanced digital signal processor (DSP) performs any computing intensive tasks required by voice and data integration. DSP processing digital signals is mainly used to perform complex calculations, otherwise these calculations may have to be performed by a general-purpose CPU. The combination of their specialized processing capabilities and low cost makes DSP well suited to perform the signal processing functions in VOIP systems

the computing overhead of G.729 Speech Compression on a single speech stream is usually large, and it is required to reach 20mips. If a central CPU is required to process multiple speech streams, it is unrealistic to also perform routing and system management functions. Therefore, using one or more DSPs can unload the computing tasks of complex speech compression algorithms from the central CPU. In addition, DSP is also suitable for voice activity detection and echo cancellation, because they process voice data stream in real time and can quickly access on-board memory, so. In this chapter, we introduce in detail how to realize the functions of speech coding and echo cancellation on the tms320c6201dsp platform. Table 1 main technical progress promoting VoIP

protocol and standard software and hardware H.323 weighted fair queuing method DSP MPLS label switching weighted random early detection advanced ASIC RTP, RTCP dual funnel universal cell rate algorithm DWDM RSVP rated access speed SONET DiffServ, car Cisco fast forwarding CPU processing power G.729, G.729A: CS-ACELP extended access tables ADSL, RADSL, SDSL frf 11/FRF. 12 token bucket algorithm multilink PPP frame relay data rectification SIP priority based cos packet over SONET IP and ATM qos/cos integration

2, advanced application specific integrated circuit (ASIC) development has produced faster, more complex and more powerful ASIC. ASIC is a special application chip that performs a single application or a small group of functions. Because they focus on very narrow application goals, they can highly optimize specific functions. Usually, dual general-purpose CPUs are one or several orders of magnitude faster. Just as reduced instruction set computer (RSIC) chips focus on fast execution of a limited number of operations, asics are pre programmed to perform a limited number of functions faster. Once the development is completed, the cost of mass production of ASIC is not high, and it is used in network devices such as routers and switches to perform functions such as routing table lookup, packet forwarding, packet classification and inspection, and queuing. The use of ASIC makes the equipment have higher performance and lower cost. They provide increased broadband and better QoS support for the network, so they play a great role in promoting the development of VoIP

3. IP transmission technology

time division multiplexing is mostly used in transmission telecommunications, and statistical multiplexing variable length packet switching is especially required. Compared with the two, the latter has high utilization rate of network resources, simple and effective interconnection and interworking, and is very suitable for data services, which is one of the important reasons for its rapid development. However, broadband IP network communication puts forward strict requirements for QoS and delay characteristics. Therefore, the development of statistical multiplexing variable length packet switching technology has attracted people's attention. At present, in addition to the new generation of IP protocol IPv6, the world Internet Engineering Task Force (IETF) has proposed multi protocol label switching (MPLS), which is a kind of label/label switching based on network layer routing, which can improve the flexibility of routing, expand the ability of network layer routing, simplify the integration of routers and cell switching, and improve network performance. MPLS can not only work as an independent routing protocol, but also be compatible with the existing network routing protocol, supporting various operation, management and maintenance functions of IP network, greatly improving the QoS, routing, signaling and other performance of IP network communication, reaching or approaching the level of statistical multiplexing fixed length packet switching (ATM), but also simpler, more efficient, cheaper and applicable than ATM. IETF also pays close attention to new packet management techniques in order to realize QoS routing. Among them, "tunneling technology" is being studied to realize broadband transmission of one-way link. In addition, how to select the IP network transmission platform is also an important field of research in recent years. IP over ATM, IP over SDH, IP over DWDM and other technologies have appeared successively. The currently recognized broadband network analysis model is shown in Figure 3

the first layer is the grass-roots foundation, which provides a high-speed data transmission backbone. The IP layer provides IP users with high-quality IP access services with certain service guarantees. The user layer provides access forms (IP access and broadband access) and service content forms. In the basic layer, Ethernet is taken for granted as the physical layer of IP network, but IP overdwdm is the latest technology and has great development potential

dense wave division multiplexing (DWDM) injects new vitality into the optical fiber network and provides amazing bandwidth in the new optical fiber backbone laid by telecom companies. DWDM technology utilizes the capabilities of optical fibers and advanced optical transmission equipment. The name of wavelength division multiplexing is derived from the transmission of multiple wavelengths of light (laser) on a single fiber. The current system can transmit and recognize 16 wavelengths, while the future system can support 40~96 full wavelengths. This is of great significance, because every additional wavelength adds an information flow. Therefore, the 2.6gbit/s (OC-48) network can be expanded 16 times without laying new optical fibers

most new optical fiber networks operate OC-192 at the speed of (9.6gbit/s), and when combined with DWDM, they produce a capacity of more than 150gbit/s on a pair of optical fibers. In addition, DWDM provides

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