asterisk users - Jun 2023 - Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

Hello,

I finally got to look at chan_sip to chan_pjsip migration again. This 
time I’m having problems with influencing codec selection on originating 
(calling) channel. It looks like PJSIP_MEDIA_OFFER only works on 
outbound (called) channel and has no affect on calling channel. My 
experiments and function documentation (which says “Media and codec 
offerings to be set on an outbound SIP channel prior to dialing.”) seem 
to confirm it.
So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s 
equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s equivalent 
of ${SIP_CODEC_INBOUND}? Or, in other words, what are we supposed to do 
to influence /calling/ channel codec selection from dialplan?
I’m working with asterisk 20.3.0.

Thank you,
Michael

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Hello,

Anyone? I have hard time to believe this is not possible with chan_pjsip.

Anyway, may I ask how people handle the following scenario which I 
imagine should be quite common:

- I have internal extensions talk to each other using g722. so their 
codec setting (with chan_sip now) is "allow=g722,ulaw"
- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between internal extensions naturally happen over g722 as its 
their preferred codec
- for external calls I now set SIP_CODEC_INBOUND=ulaw to influence codec 
selection on calling channel and the calls set up using ulaw end-to-end

Can somebody please advise how to achieve the same with chan_pjsip?

Thanks,

Michael

On 6/30/23 09:30, Michael Ulitskiy wrote:
>
> Hello,
>
> I finally got to look at chan_sip to chan_pjsip migration again. This 
> time I’m having problems with influencing codec selection on 
> originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only 
> works on outbound (called) channel and has no affect on calling 
> channel. My experiments and function documentation (which says “Media 
> and codec offerings to be set on an outbound SIP channel prior to 
> dialing.”) seem to confirm it.
> So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s 
> equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s 
> equivalent of ${SIP_CODEC_INBOUND}? Or, in other words, what are we 
> supposed to do to influence /calling/ channel codec selection from 
> dialplan?
> I’m working with asterisk 20.3.0.
>
> Thank you,
> Michael
>
>
>
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