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

[Michael Maier]

Hello Michael,

you are referring to the following behavior - did I get it correctly?:

outbound broken: asterisk offers g722 / g711 to provider (callee), 
callee answers g711. Asterisk now transcodes between caller and callee 
(g722 <-> g711).

inbound works: call from provider: g711 -> asterisk drops g722 and 
passes g711 to internal callee -> no transcoding.


As far as I know, there is no working solution as of now. I discussed 
this problem years ago already here but unfortunately nothing usable 
happened so far (which I would know off). The priority is not high 
enough. I need a solution, too. I understand that this behavior is a 
nogo if you have a lot of calls because transcoding is expensive.


Thanks
Michael



On 05.07.23 at 17:58 Michael Ulitskiy wrote:
> 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
>