[asterisk-dev] Advanced Codec Negotiation: Need info and uses cases

[Michael Maier]

On 25.06.20 at 15:57 George Joseph wrote:
> On Wed, Jun 24, 2020 at 12:03 PM Stefan Tichy <asterisk3 at pi4tel.de> wrote:
> 
>> On Sat, Jun 06, 2020 at 08:20:47AM +0200, Michael Maier wrote:
>>
>>> Transcoding only if there is an allowed and supported codec on both sides
>>> which are not common. If there is one common codec on both sides - use
>>> this codec and do not transcode.
>>
>> If Alice (OpenStage20, Local Net) offers G.722, alaw and ulaw and
>> Bob (Remote Localtion) supports Opus, alaw and ulaw, then
>> transcoding between G.722 and Opus results in better audio quality.
>> Transcoding causes some CPU Load, but this might be acceptable.
>>
>>
> Hmmm.  I'll have to think if we can support this generically.
> It'd need an option like
> "use_the_highest_bitrate_codec_even_if_it_means_transcoding" :)

I doubt that this is a good thing. Why should adding additional delay raise quality? opus is lossy. Each transcode step adds errors. It can't be better.

Please think about cascades, too! The first one transcodes opus -> g722, the next step g722 -> opus, ... . Great.

But it would explain, why some fax calls are horribly broken if you think a mechanism like this further.


Thanks
Michael