The traditional telephone network transmits voice by circuit switching, and the required transmission bandwidth is 64kbit / s. The so-called VoIP is based on the IP packet switching network as the transmission platform, which compresses and packages the analog voice signal, so that it can be transmitted using the connectionless UDP protocol.

　　In order to transmit voice signals over an IP network, several elements and functions are required. The simplest form of network consists 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. Figure 1 shows how VoIP devices convert voice signals into IP data streams and forward these data streams to IP destinations, which in turn convert them back to voice signals. The voice of the two networks must support IP transmission, and can be any combination of IP router and network link. Therefore, the transmission process of VoIP can be simply divided into the following stages. 　　

Figure 1 model structure of VoIP 

　　1. Voice to data conversion

　　Voice signal is an analog waveform. Voice is transmitted through IP. Whether it is real-time application service or non real-time application service, Dao Maoan first needs to 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. 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. The cost of voice transmission is usually 240 ms or 60 ms. Digitization can be realized by various speech coding schemes, and the current 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 speech signal is digitally encoded, the next step is to compress the speech packet with a specific frame length. Most encoders have a specific frame length. If an encoder uses 15 ms frames, the first 60 ms packet is divided into four frames and encoded in sequence. Each frame contains 120 speech samples (the sampling rate is 8kHz). After encoding, four compressed frames are combined into a compressed voice packet and sent to the network processor. The network processor adds packet header, time scale and other information to the voice and transmits them to the other end point through the network. Voice network simply establishes a physical connection (a line) between communication endpoints and transmits coded signals between them. Unlike circuit switching network, IP network does not form a connection. It requires data to be put 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, the whole network is regarded as a receiving voice packet from the input and then transmitting it to the output of the network within a certain time (T). T can change in a whole range, which reflects 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 destination path. The network link can be any topology or access method supporting IP data flow.

　　4. IP packet data conversion

　　The destination VoIP device receives the IP data and starts processing. The network level provides a variable length buffer to adjust the jitter caused by the network. The buffer can hold many voice packets, and users can choose the size of the buffer. Small buffers produce less delay, but cannot adjust large jitter. Secondly, the decoder decompresses the encoded voice packet to generate a new voice packet. This module can also operate according to the frame, which is the same length as the decoder. If the frame length is 15ms, the 60ms voice packet is divided into four frames, and then they are decoded and restored to 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 data is retained, and then the original data is provided to the decoder.

　　5. Digital voice to analog voice

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

Figure 2 basic process of VoIP transmission 

　　The more branches or partners an enterprise or organization has, the more necessary it is for departments to communicate with each other. If more and more customers of an enterprise feel that they are not suitable for one branch, the number will be released. Moreover, the branches of most enterprises in different regions must have different division of labor and different advantages. How to strengthen the internal humanized communication between various departments of enterprises or institutions? How to make customers feel that the enterprise is an orderly and easy to communicate organization, rather than a scattered organization that is difficult to communicate? How to make use of different backgrounds and advantages? These problems are the most concerned problems of enterprise management.

　　Program summary:

　　Scheme summary: the maturity of network technology provides a good foundation for solving these problems. So can sending and receiving email or simply sharing data resources through the network solve the problem? The answer is yes, absolutely not. On the one hand, not all employees in any company are proficient in it. In this era of increasingly fragmented career, staff in different departments have different majors and different business contents to learn. And some of the things transmitted in the network can never be as clear and concise as the language expression.

　　So, can it work to rent the function of the program-controlled exchange of Telecom, or to set up a call center? Of course, it is certainly possible to rent nationwide, make use of the functions of Telecom telephone exchanges, or set up call centers across regions. However, the high rental cost is bound to increase the operating costs of enterprises, which enterprises or institutions certainly do not want to see. So is there any cheap and practical solution?

　　So, can it work to rent the function of the program-controlled exchange of Telecom, or to set up a call center? Of course, it is certainly possible to rent nationwide, make use of the functions of Telecom telephone exchanges, or set up call centers across regions. However, the high rental cost is bound to increase the operating costs of enterprises, which enterprises or institutions certainly do not want to see. So is there any cheap and practical solution? Naturally, the answer is yes. VoIP (voice over IP) technology provides the best solution for the above problems. It uses IP technology to transmit voice with a bandwidth of less than 10k. Moreover, with the extensive application of DSP (digital signal processing) chips and the cooperation of flow control, VoIP gateway can solve many difficult problems for you, and let you care about your specialty and your field.

　　Features of the scheme:

　　As shown in the right figure, take gateplus voice gateway as an example:

　　1. It can provide free calls from branch to branch, such as free calls from extension A1 to extension B1 and extension C1.

　　2. Realize free call from network telephone to each extension. Such as network telephone 1 to extension A1, B1, etc.

　　3. Realize the free call between the computer running the network telephone software and each extension and the network telephone.

　　4. Use the skip dial function, combined with local telephone, to realize the communication between VoIP and local telephone. As shown in the figure, for the call from extension A1 to local telephone C or mobile telephone C, only the local telephone fee at local telephone C or local mobile telephone fee at mobile telephone C is charged.

　　5. Use the skip dial function, combined with local telephone, to realize the communication between VoIP and local telephone. As shown in the figure, for the call from local telephone a to local telephone C or mobile telephone C, only the local telephone fee of local telephone a and local telephone C or the local mobile telephone fee of mobile telephone C will be charged.

　　6. Gateplus series VoIP products also have more powerful functions such as hotline, embedded gatekeeper (some models), voice enhancement, etc