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

Michael Ulitskiy mulitskiy at acedsl.com
Thu Jul 6 12:46:22 CDT 2023


FYI: i've created a feature request to add SIP_CODEC_INBOUND equivalent 
functionality to chan_pjsip:

https://github.com/asterisk/asterisk-feature-requests/issues/9

Let's see where it goes

*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https://www.aceinnovative.com
On 7/5/23 11:58, 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
>
> 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
>>
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230706/274a7275/attachment.html>


More information about the asterisk-users mailing list