What are codecs?

Overview of audio codecs used in sipgate VoIP telephony, including supported codecs and data usage.

In order to transmit acoustic signals from a transmitter to a receiver, they must first be converted into data. This task is performed by a codec. A codec is a coherent pair of algorithms that is responsible for encoding and decoding data. For larger amounts of data, a codec is also responsible for compressing and decompressing the data. Codecs play an important role in VOIP telephony, as they control the quality of the connection.

Compressing codecs can be divided into two groups, lossy and lossless codecs. Lossy codecs can achieve higher compression of the data packets and therefore require less bandwidth during transmission. However, this has a negative impact on quality. Lossless codecs transmit the data packets without compression and therefore without loss of quality, but require a higher bandwidth.

Which codecs are supported with sipgate?

• G.711a (sipgate VoIP/SIM/trunking)

• G.711u (sipgate VoIP/SIM/trunking)

• GSM (sipgate VoIP/SIM/trunking)

• G.722 (sipgate VoIP/SIM/trunking)

• OPUS (satellite and sipgate App)

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Which RTP / Sprachcodecs are used by sipgate?

• G.711: approximately 100 kbit/s

• GSM: 13 - 20 kbit/s

• G.722/ HD

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Which codecs are used by satellite and sipgate App?

• Opus: 6kbit/s - 510kbit/s

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How much data is used during a standard call?

• VoIP call via G.711 / 10 Min. approximately 7,5MB

• VoIP call via GSM / 10 Min. approximately 0,975-1,5MB

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If you are experiencing issues with lost audio or with speech quality please see this article.