The Basic Transmission Process of VoIP

 The traditional telephone network transmits voice by circuit exchange, and the required transmission broadband is 64 k bit/s. The so-called VoIP is based on IP packet switching network as the transmission platform, The analog voice signal is compressed, packaged and a series of special processing, so that it can use the connection-less UDP protocol for transmissionl.

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 analog waveform, through the IP way to transmit voice, Whether it’s a real-time application or a non-real-time application.Firstly, the speech signal should be converted to analog data, that is, the analog speech signal should be quantized by 8 or 6 bits, and then sent to the buffer storage area, the size of the buffer can be selected accordingto the delay and coding requirements. Many low bit rate encoders are based on frame coding.

Typical frame lengths range from 10 to 30ms. Considering the cost during transmission, the interspeech packet usually consists of 60, 120 or 240ms of voice data.Digitization can be achievedby using various speech coding schemes, the main one being ITU-T G.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 packet headers, timestamps, and other information to the voice and sends it over the network to the other endpoint.

The voice network simply establishes physical connections (a line) between the communication endpoints and transmits the encoded signals between the endpoints. Unlike circuit-switched networks, IP networks do not form connections; instead, they require data to be placed in variable-length datagrams or packets, each of which is then sent over the network with addressing and control information and forwarded from station to station to its destination

3. Transfer
In this channel, the whole network is seen as receiving a voice packet from the input and then delivering it to the network output within a certain time (t). t can vary in some full range, reflecting jitter in network transmission.
Peers in the network examine the addressing information attached to each IP packet and use this information to forward the datagram to the next station on the path to its destination. A network link can be any topology or access method that supports IP data flows.

4、 IP package- data conversion

The destination VoIP device receives this IP data and starts processing. The network level provides a variable length buffer used to regulate the jitter generated by the network. The buffer canaccommodate 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.During the processing of the datagram, the addressing and control information is removed, and the original raw data is preserved, which is then provided to the decoder.

5、 Digital voice conversion to analog voice

The playback driver takes out the speech sample points (480) in the buffer and sends them to the sound card, and broadcasts them through the speaker at a predetermined frequency (for example, 8kHz). In a nutshell, the transmission of voice signals over IP networks goes through the conversion from analog signals to digital signals, the encapsulation of digital voice into IP packets, the transmission of IP packets through the network, the unpacking of IP packets, and the restoration of digital voice to analog signals.

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