Re: Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

Well, I'm trying to migrate to chan_pjsip so that I don't have to do that.

It's so surprising that the issue so seemingly obvious and trivial 
hasn't been addressed yet that I wanted to query the collective wisdom 
of this list to verify my observations.

Thanks for github pointer.

Michael

On 7/5/23 16:46, [email protected] wrote:
> On 7/5/2023 4:19 PM, Michael Ulitskiy wrote:
>>
>> Hi Michael,
>>
>> Thanks for the reply.
>>
>> I was referring to the scenario you named as 'outbound broken'. I 
>> didn't get to look at inbound call behavior yet, as I got stuck with 
>> inability to avoid transcoding on outbound calls.
>>
>> To be more specific the scenario is as follows:
>>
>> 1. a phone initiates a call offering g722,g711 to asterisk
>> 2. asterisk creates outbound call to carrier offering g711 only 
>> (carrier only supports g711)
>> 3. carrier accepts the call and outbound call leg is now running on g711
>> 4. asterisk accepts a phone's call with g722 since it's allowed on 
>> phone's endpoint and was indicated as preferred in phone's INVITE and 
>> now initial call leg is running on g722, resulting in transcoding
>>
>> This is very disappointing. Since developers announced their plans to 
>> drop chan_sip from future asterisk versions
>>
> It's already been removed and won't be in any future major releases.
> If you still need chan_sip after removal, you can continue adding it 
> from out of tree and building it. I maintain a working version of it 
> out of tree.
>>
>> I was under impression that chan_pjsip has reached feature paritiy 
>> with chan_sip.
>>
> It has mostly, but not completely, no.
>>
>> What is needed is an ability to tell asterisk which codecs are 
>> allowed to be included in "200 OK" asterisk sends back to the phone. 
>> I guess we need to submit a feature request. How do we go about it 
>> these days?
>>
> I'm not sure about the particulars of this issue at all, but to answer 
> the question at hand, there's a repo for it: 
> https://github.com/asterisk/asterisk-feature-requests.

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