Raise mixing sample rate

Asterisk Asterisk Support
#1

Whenever I have a conference call the audio comes in as opus@48000 and then gets converter down to slin@8000 for mixing just to be converted back up to opus@48000. My concern with this is the audio quality is much lower when this happens in conference calls than during direct calls when opus@48000 is sent directly.

I am wondering if there is any place to change it so that it mixes at opus/slin@48000 instead of degrading to slin@8000? I have tried changing “internal_sample_rate” to 48000 in confbridge.conf but have seen no change. I have gone through all the sample config files but can’t find a reference to the default conference format.

#2

It dynamically adjusts based on what channels are in the conference bridge. What is the specific assortment of channels and such?

#3

I’ve got one channel coming from a device and another channel from software listening to the device in the conference.

They are both on PJSIP, the device being listened to has ReadTranscode opus@48000>slin@48000>slin@8000 and the listening software has the WriteTranscode opus@48000>slin@48000>slin@8000.

I am expecting that these would change with the mixing rate adjustment. Is there somewhere I should be monitoring this?

#4

Enabling core debug and looking at the output of ConfBridge may show what is going on.

#5

I don’t see much output. Since I am doing a one-way call is it possible to simply broadcast the audio as opus instead without degrading to slinear in a conference?

#6

Are you using Originate and local channels?

#7

Both channels are coming through PJSIP, I am not certain where to check if they are using Originate or local channels.

#8

I can confirm now that no local channels are being used

#9

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