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
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