[asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

[Michael Ulitskiy]

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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