Which SIP codec should I use?

Copied from: https://teliax.zendesk.com/entries/167709-which-codec-should-i-use

The amount of bandwidth used for each call depends mostly on your selection of codec. The following codecs and their approximate bandwidth utilization (including overhead) are supported by Teliax:

- G.722: Wideband audio at data rates from 48, 56 and 64 kbit/s (HD Voice)
- G.711u: 64 kbit/s
- G.726: 16, 24, 32, and 40 kbit/s
- G.729a: 8kbit/s
- GSM: between 6.5 (half-rate) and 13 kbit/s (full-rate)
There are many strengths to the codecs we support. The decision for which codec to use depends largely on your network. For example, G.711u will provide you with PSTN grade call quality but at a higher bandwidth cost. G.729 will provide you with almost the same quality at a much lower hit to your bandwidth, however there is a per channel license required. Things to consider are cost, availability of bandwidth, and the capabilities of the equipment or software you are using.

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