there is very nice article about opus in freeswitch from Giacomo Vacca
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/12517398
<https://freeswitch.org/confluence/plugins/servlet/mobile#profile/giavac>
create something similiar for Opus in Asterisk is good objective for
asterisk opus working group
any news about how to setup working group? (or get write access to
wiki.asterisk.org?)
Dne 31/10/2016 v 22:20 Alexander Traud napsal(a):
Has anybody updated the version of the [Opus transcoding] patch for 14 and/or
master?
<https://github.com/traud/asterisk-opus>
The story continues there. I updated the code and I try my best to maintain
that fork (because I use it myself). Currently, it should be compatible from
13.7 up to the latest 14 and Master version. If not, please, create an Issue
Report or a Pull Request. Other contributions are welcome as well.
it does take a small bits of maintenance
If there is anything I could do, to ease that, please do not hesitate.
Opus is to become the new standard audio codec.
I know, you want to persuade Digium to give more attention to that audio codec
and its features (Native PLC, Adaptive FEC, VAD/DTX/CNG). Yes, those features
of a Media Gateway are important not only to Opus but other audio codecs as
well, like 3GPP EVS. Actually, because of the complexity of modern audio
codecs (since the end of the nineties with the introduction of G.729), such
features are a must-have. Without those features, distortion is a common issue
with modern audio codecs.
Furthermore, I cannot restrain to comment on that statement: I am quite
skeptical about the future of Opus. Currently, it is there because of WebRTC.
Full stop! Mobile phones go for GSM, AMR, AMR-WB, and last years 3GPP EVS.
Landline phones go for and continue to use G.711, G.726-32, G.722. If Opus gets
lucky, the industry chooses Opus for multi-channel and music via landline.
However last year, for music, the German company AVM went not for Opus but for
its precursor CELT.
Finally, from my experiences with several implementations, Opus Codec seems to
be quite challenging. Not many implementations leverage the parameter
negotiation via the SDP attribute fmtp. It was a bit of work to add that to
Asterisk, by the way. Therefore sometimes, tailoring of the Opus Codec is
impossible and the data rate goes through the sky. When it comes to wide-band
audio codecs, G.722 and AMR-WB might stay the winners because they are more
limited.
Nevertheless, I would have nothing against a single audio codec for VoIP/SIP -
finally.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
