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
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