Transferring files via the VoIP-telephony is carried out by means of encoding and receipt of data in a digital format. The process involves the ADC installed in the IP PBX. The analog signal is recovered from the sampling of the frequency that is greater than the maximum in the signal spectrum.
Encoding involves the compression of the digital information material, with its subsequent decoding. This distortion of the sound should be minimal because it affects the quality of communication.
For this reason, many experts are interested in the question of choice of the speech codec since the improper approach to configuring the gateway VoIP is fraught with not the system failing to start and interruptions in its work.
There are many codecs. The most popular of them are G711, G729, G723.1, GSM (RPE-LTP), and iLBC (Internet low bitrate codec). Data transmission depends on the type of equipment used. GSM (RPE-LTP) functions when connecting the GSM and VoIP, and the G711 is used for ISDN devices and used by the system as the primary digital channel distributed in the joint application with SIP and SCCP. The standard applies Mu-law and A-law. Mu-law is in common use in North America, A-law in Europe.
Depending on the bandwidth, the sound quality varies when using different codecs. Accordingly, G711 should be used at a rate of at least 64 KB/s. If the data rate is low - from 8 Kb/s, then it is better to give preference to G729. This compressing codec provides for the possibility of losing a small part of the data. It transmits the digitized signal parameters that are sufficient for speech synthesis, while retaining all the characteristics of the voice. It is believed that the G723.1 codec is not able to transmit sound properly. G711 or iLBC (Internet low bitrate codec) are suitable for better transmission.
Bandwidth codecs are arranged as follows:
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The first one is characterized by the lowest bandwidth and the last one the highest. Based on it, this codec should be selected and determined. Codecs that are able to work via with the high-speed Internet only cannot transmit signals without serious distortion if the Internet connection is poor. Therefore, their quality grading is as follows:
The first one indicates a better voice quality, and then - in descending order.
When choosing a codec, the system selects a suitable solution for a better way to transmit voice. Due to the best quality, G.711 is selected for the fax transmission that requires excellent throughput, while G.729 for GSM, Frame Relay voice data transmission, used with CS-ACELP technology. Its transfer rate is up to 30 KB/sec. G.729 is supported by any VoIP device. At the same time, G.722 gradually supersedes, G.711. G.722 provides the best possible call quality. It is used in Cisco 79xx, Grandstream GXP20xx, Polycom 5xx and 6xx, and Snom 3xx.
This short review of the codecs will allow users to take into account the necessary nuances when setting up the Gateway GSM ip, avoiding blunders and failures in the work of equipment.
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