[asterisk-users] Opus Codec - real world adoption

[marek]

Hi,

today i made some tests with Opus codec (webrtc app, asterisk + pjsip + wss)

Some questions arose.

Sangoma questions

- Can Sangoma publish usage data from codec_opus anonymous statistic?

- Any plans port codec_sangoma to Opus 1.3? current is 1.2 based on 
https://issues.asterisk.org/jira/browse/ASTERISK-29580

- Any plans to open source code and integrate to Asterisk as other codecs?


Public questions

- Any experience with wide usage of Opus for endpoints transcoded to 
alaw/ulaw for SIP trunk to telco provider? (i.e CPU  increase percentage 
compared to only alaw scenario)

- Any experience with "better" audio on slow/problematic internet 
connection?

- Any experience with call recordings on endpoint side(pbx user) and 
speech to text transcription? (compared to calls in alaw/ulaw)

Thank you

Have a nice weekend

Marek Cervenka