Asterisk 20, PJSIP and ACN - what's the status?

[lordgurke]

Hi,

I’m currently running Asterisk 20.4.0 and try to get Advanced Codec Negotiation working.
Just as a quick check: Is the ACN implementation working to do the following?

I want to make achieve, that local clients can call each other using OPUS but when dialling outside, should use G.722 or G.711a - so I basically want to avoid unnessecary transcoding.

Endpoint config for local phones:

allow: !all,opus,g722,alaw
codec_prefs_outgoing_answer = prefer: pending, operation: intersect, keep: first, transcode: prevent
codec_prefs_outgoing_offer = prefer: pending, operation: intersect, keep: first, transcode: prevent

Endpoint config for my SIP trunk:

allow: !all,g722,alaw
codec_prefs_incoming_answer = prefer: pending, operation: intersect, keep: first, transcode: prevent
codec_prefs_incoming_offer = prefer: pending, operation: intersect, keep: first, transcode: prevent

This has no effect at all. Calls still get transcoded from Opus to G.722.
I fiddled with the settings in many ways but was unable to get this working.
From what I see from debugging, it seems that the whole codec_prefs_* settings seem to be ignored when making a call, thus being shown in pjsip show endpoint.

Just to make sure: Is this feature implemented in Asterisk 20 or am I making some wrong settings?

[silentindark]

FYI → [new-feature]: Advanced Codec Negotiation · Issue #223 · asterisk/asterisk · GitHub

[jcolp]

There is an associated pull request currently targeted for the next major standard release. It is possible it will show up earlier, if someone makes the changes required to have it go into earlier versions (as written it can break existing working configurations).

[lordgurke]

Oh, then I’ll be a bit more patient to wait for it
Thanks!