The traditional telephone network transmits voice by circuit exchange, and the required transmission broadband is 64 k bit/s. The so-called VoIP is the IP packet switching network as the transmission platform, to simulate the simulated voice signal compression, packaging and a series of special processing, so that it can be transmitted by the unconnected UDP protocol.
Several elements and functions are required to transmit voice signals on an IP network. The simplest form of the network consists of two or more devices with VoIP capabilities that are connected through an IP network.
1. Voice-to-data conversion
Voice signal is simulated waveform, through IP voice, whether real-time application business or real-time application business, the first to simulate voice signal data conversion, which is the simulated voice signal for eight or six quantification, and then into the buffer storage area, the size of the buffer can be selected according to the requirements of the delay and coding. Many low bit-rate encoders are taken to encode in frame units.
Typical frame length ranged from 10 to 30 ms. Considering the costs in transmission, interlingual packets usually consist of 60,120 or 240ms of speech data. Digitization can be realized using various voice coding schemes, and the current speech coding standards are mainly ITU-TG.711. The voice encoder at the source destination must implement the same algorithm so that the speech device at the destination can restore the analog speech signal.
2. the original data to the IP conversion
Once the speech signal is digitally encoded, the next step is to compress and encode the speech packet with a specific frame length. Most of the encoders have a specific frame length. If an encoder uses a 15ms frame, the 60ms package from the first one is divided into four frames and encoded in order. Each frame has 120 speech samples (sampling rate 8 kHz). After encoding, the four compressed frames are synthesized into a compressed speech packet and sent into the network processor. The network processor adds Baotou, time mark and other information to the voice and transmits it to the other end point through the network.
The voice network simply establishes physical connections (a line) between the communication endpoints and transmits the encoded signals between the endpoints. Unlike a circuit switching network, an IP network does not form a connection. It requires the data to be placed in a variable length datagram or packet, then addressed and control information to each datagram, and sent through the network, forwarded to the destination.
3. Transfer
In this channel, the entire network is seen as a voice packet received from the input side and then transmits it to the network output within a certain time (t). The t can vary in a full range, reflecting the jitter in the network transmission.
The same node in the network checks the addressing information associated with each IP data and uses this information to forward that datagram to the next stop on the destination path. A network link can be any topology or access method that supports IP data flow.
4. The IP package- -the transformation of the data
The destination VoIP device receives this IP data and begins the processing. The network level provides a variable length buffer used to regulate the jitter generated by the network. The buffer can accommodate many voice packets, and users can choose the size of the buffer. Small buffers produce smaller delays but cannot regulate large jitter. Secondly, the decoder unpresses the encoded speech package to produce a new speech package. This module can also be operated by frame, which is exactly the same length as the decoder.
If the frame length is 15ms, the 60ms speech packets are divided into 4 frames, and then they are decoded to a 60ms speech data stream and sent into the decoding buffer. In the processing process of the data report, the addressing and control information are removed, the original original data is retained, and then the original data is provided to the decoder.
5. Digital speech is converted to analog speech
The playback drive removes the voice sample points (480) from the buffer and sends them to the sound card through the speaker at a predetermined frequency (e. g., 8 kHz). In short, the transmission of voice signals on the IP network should go through the process of conversion from analog signal to digital signal, digital voice packaging into IP packet, transmission of IP packet through the network, IP packet unpacking and restoration of digital voice to analog signal.
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