Why the Glass wall Curtain LED display can lead the fashion of outdoor media?
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Basic transmission process
The traditional telephone network of VoIP is to transmit voice by circuit switching, and the required transmission bandwidth is 64kbit / s. The so-called VoIP is based on the IP packet-switched network as a transmission platform, which compresses and packs analog voice signals and a series of special processing, so that it can be transmitted using the connectionless UDP protocol.
In order to transmit voice signals on an IP network, several elements and functions are required. The simplest form of network consists of two or more VoIP-enabled devices, which are connected via an IP network. How VoIP equipment converts voice signals into IP data streams, and forwards these data streams to IP destinations, and IP destinations convert them back to voice signals. The voice of the two networks must support IP transmission, and can be any combination of IP routers and network links. Therefore, the transmission process of VoIP can be simply divided into the following stages.
Principles of VoIP technology 1. Voice-to-data conversion
The voice signal is an analog waveform, and the voice is transmitted through the IP method. Whether it is a real-time application service or a non-real-time application service, the analog data conversion of the voice signal must be performed first, that is, the analog voice signal is quantized by 8 or 6 bits. Sent to the buffer storage area, the size of the buffer can be selected according to the requirements of delay and encoding. Many low bit rate encoders use frames as the unit for encoding. The typical frame length is 10 ~ 30ms. Considering the cost in the transmission process, inter-language packets usually consist of 60, 120, or 240 ms of voice data. Digitization can be achieved using a variety of speech coding schemes. The currently adopted speech coding standards are mainly ITU-T G.711. The source and destination voice encoders must implement the same algorithm so that the voice device at the destination can restore the analog voice signal.
Principle of VoIP technology 2. Conversion of original data to IP
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 a 15ms frame, the 60ms packet from the first will be divided into 4 frames and encoded in order. Each frame contains 120 speech samples (sampling rate is 8kHz). After encoding, the 4 compressed frames are combined into a compressed voice packet and sent to the network processor. The network processor adds packet headers, time stamps and other information to the voice and transmits it to another endpoint through the network. The voice network simply establishes a physical connection (a line) between the communication endpoints, and transmits the encoded signal between the endpoints. Unlike a circuit-switched network, an IP network does not form a connection. It requires placing data in variable-length datagrams or packets, and then attaching addressing and control information to each datagram and sending it through the network, one station at a time. Station-to-site forwarding.
Principles of VoIP technology 3. Transmission
In this channel, the entire network is regarded as a voice packet received from the input terminal, and then transmitted to the network output terminal within a certain time (t). t can vary within a certain range, reflecting jitter in network transmission. The peer nodes in the network check the addressing information attached to each IP data and use this information to forward the datagram to the next stop on the destination path. The network link can be any topology or access method that supports IP data streams.
VoIP technology principle 4. IP packet-data conversion
The destination VoIP device receives this IP data and starts processing. The network level provides a variable length buffer to adjust the jitter generated by the network. The buffer can hold many voice packets, and the user 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 be operated frame by frame, exactly the same length as the decoder. If the frame length is 15ms, 60ms voice packets are divided into 4 frames, and then they are decoded and restored to 60ms voice data stream and sent to the decode buffer. During the processing of the datagram, the addressing and control information is removed, the original original data is retained, and then the original data is provided to the decoder.
Principle of VoIP technology 5. Convert digital voice to analog voice The playback driver takes out the voice samples (480) in the buffer and sends them to the sound card, and broadcasts them through the speaker at a predetermined frequency (for example, 8 kHz).
In short, the transmission of voice signals on the IP network goes through conversion from analog signals to digital signals, digital voice is encapsulated into IP packets, IP packets are transmitted through the network, IP packets are unpacked, and digital voice is restored to analog signals Wait for the process.
Principles of VoIP technology for IP phones
In the current network communication, Email service is not the preferred communication method now. More instant messaging, voice services, etc. are emerging one after another on the network. So, now let ’s talk about the principles of VoIP technology for IP phones.