asterisk users - Jul 2023 - 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, asterisk at phreaknet.org 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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I suspect most people simply don't care.   Transcoding between ulaw and 
g722 is not CPU intensive and Direct Media doesn't work when NAT is 
involved (which would the case for most people).

On 7/5/23 17:22, Michael Ulitskiy wrote:
> 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.