[asterisk-dev] pjsip advanced codec negotiation

[Trevor Peirce]

Hi All,

Trying to wrap my head around codec negotiation with Asterisk 18 when 
using PJSIP.  I'm not seeing the results that I expect.

I have the following configuration:

Phone:
  allow                              : (g722|ulaw)
  codec_prefs_incoming_answer        : prefer:pending, 
operation:intersect, keep:all, transcode:allow
  codec_prefs_incoming_offer         : prefer:pending, 
operation:intersect, keep:all, transcode:allow
  codec_prefs_outgoing_answer        : prefer:pending, 
operation:intersect, keep:all, transcode:allow
  codec_prefs_outgoing_offer         : prefer:pending, operation:union, 
keep:all, transcode:allow
  incoming_call_offer_pref           : remote_first
  outgoing_call_offer_pref           : remote

The phone itself generates SDP with the following order OPUS, G722, 
G711u, and SILK.

When I place a call that doesn't go to another endpoint, for example to 
VoiceMailMain, the call is answered in ulaw and then Asterisk 
immediately reinvites to g722.

I'd prefer to just have Asterisk answer g722 directly and not generate a 
reinvite.

Any pointers would be greatly appreciated.

Thanks,

-- 
Trevor Peirce
AcroVoice Solutions Inc

www.acrovoice.ca - 1-888-606-3030 x701