> For the complete documentation index, see [llms.txt](https://help.sipgate.de/llms.txt). Markdown versions of documentation pages are available by appending `.md` to page URLs; this page is available as [Markdown](https://help.sipgate.de/cloud-telefonanlage/en/using-sipgate/telefonie/was-sind-codecs.md).

# What are codecs?

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

To be able to transmit acoustic signals from a transmitter to a receiver, they must first be converted into data. This task is performed by a **Codec** handled by. A codec is a connected pair of algorithms that is responsible for the **encoding and decoding** of data. For 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 greater compression of the data packets and therefore require less bandwidth during transmission. However, this negatively affects the quality. **Lossless codecs** transmit the data packets without compression and therefore 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)

***

* **Which RTP / voice codecs does sipgate use?**
* G.711: approximately 100 kbit/s
* GSM: 13 - 20 kbit/s
* G.722/ HD

***

#### **Which codec does** [**satellite**](https://www.satellite.me/) **and** [**sipgate App**](https://clinq.com/)**?**

* Opus: 6 kbit/s - 510 kbit/s

***

#### **How much data is used in approx. ten minutes?**

* VoIP call via G.711 / 10 min. approximately 7.5 MB
* VoIP call via GSM / 10 min. approximately 1.5 MB

***

#### **Are data channels ISDN (X.75) or analog X.25 supported?**

No, not planned at the moment.


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