Audo File format sln48

hi

i’m using Sangoma Connect the desktop version, when i make phone calls and an audio file starts it is played in sln48 format

When loading the original .wav file, in System Recording Tab, is converted into every possible format: alaw, g722, g729, gsm, sln, sln16, sln48, ulaw and of course wav

However even if i try a call from a mobile phone that should go through with gsm codec, i note that the file is always passed in sln48 format which is always the heaviest, from a 472 Kb wav file you get a sln48 file of almost 3 MB

Is it normal that asterisk always uses this codec? in the sip settings the order is opus, alaw and gsm with Freepbx17 and Asterisk 22.3

What do you mean by this exactly? It is unlikely it would negotiate at gsm.

If opus is possible then it would likely have negotiated using it, which operates at 48kHz. You can see what a channel is using by invoking “core show channel” in the Asterisk CLI.

Assuming you are talking about standard mobile telephony here, rather than voice over mobile data, the air interface codec was never likely to be used beyond the base station. Also, GSM and its variants are increasingly unlikely to be used, as people demand better voice quality, and it is more likely that a speech codec covering up to 6kHz will be used.

The codec used by SIP is typically negotiated between user agents, without regard to the total path, so, if you offer Opus, as first choice, and the ITSP supports it, they are likely to use it as the preferred codec. Often only the preferred one is used.

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