What are codecs?

Audio codecs explained in sipgate – which codecs are supported and how they affect voice quality.

To be able to transmit acoustic signals from a sender to a receiver, they must first be converted into data. This task is carried out by a codec is handled by. A codec is a closely related pair of algorithms responsible for the encoding and decoding of data. With larger amounts of data, a codec also ensures the compression and decompression of the data. Codecs play an important role in VoIP telephony, as they control the quality of the connection here.

Compressing codecs can be divided into two groups: lossy and lossless codecs. Lossy codecs can achieve higher compression of data packets and therefore require less bandwidth during transmission. However, this negatively affects quality. Lossless codecs transmit the data packets without compression and thus without loss of quality, but require more bandwidth.

Which codecs are supported by sipgate?

• G.722 (approx. 100 kbit/s)

• OPUS (approx. 40 kbit/s)

• G.711a (approx. 100 kbit/s)

• G.711u (approx. 100 kbit/s)

• GSM (approx. 20 kbit/s)

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• Which RTP / voice codecs does sipgate use?

• G.711: approximately 100 kbit/s

• GSM: 13 - 20 kbit/s

• G.722/ HD

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Which codec do satellite and sipgate app?

• Opus: 6 kbit/s - 510 kbit/s

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What amounts of data are used in about ten minutes?

• VoIP call via G.711 / 10 min. approximately 7.5 MB

• VoIP call via GSM / 10 min. approximately 1.5 MB

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Are data channels ISDN (X.75) or Analog X.25 supported?

No, not planned at the moment.