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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