Opus Codec – Real World Adoption
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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One thought on - Opus Codec – Real World Adoption
I will see if this is possible, but no promises.
It is in queue but no timeframe on such a thing.
Not as of this time, no.