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<p><span style="font-size:10pt">Hi Michael,</span></p>
<p><span style="font-size:10pt">Thanks for the reply.</span></p>
<p><span style="font-size:10pt">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.<br>
</span></p>
<p><span style="font-size:10pt">To be more specific the scenario is
as follows:</span></p>
<span style="font-size:10pt">1. a phone initiates a call offering
g722,g711 to asterisk</span><br>
<span style="font-size:10pt">2. asterisk creates outbound call to
carrier offering g711 only (carrier only supports g711)</span><br>
<span style="font-size:10pt">3. carrier accepts the call and
outbound call leg is now running on g711</span><br>
<span style="font-size:10pt">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<br>
</span>
<p><span style="font-size:10pt">This is very disappointing. Since
developers announced their plans to drop chan_sip from future
asterisk versions I was under impression that chan_pjsip has
reached feature paritiy with chan_sip. 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?</span></p>
<span style="font-size:10pt">Thanks,</span><br>
<span style="font-size:10pt">Michael<br>
</span>
<div class="moz-cite-prefix"><br>
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<div class="moz-cite-prefix">On 7/5/23 14:59, Michael Maier wrote:<br>
</div>
<blockquote type="cite"
cite="mid:4e4ab8c8-c086-f1a0-6a30-49e8d7459a86@mailbox.org">Hello
Michael,
<br>
<br>
you are referring to the following behavior - did I get it
correctly?:
<br>
<br>
outbound broken: asterisk offers g722 / g711 to provider (callee),
callee answers g711. Asterisk now transcodes between caller and
callee (g722 <-> g711).
<br>
<br>
inbound works: call from provider: g711 -> asterisk drops g722
and passes g711 to internal callee -> no transcoding.
<br>
<br>
<br>
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.
<br>
<br>
<br>
Thanks
<br>
Michael
<br>
<br>
<br>
<br>
On 05.07.23 at 17:58 Michael Ulitskiy wrote:
<br>
<blockquote type="cite">Hello,
<br>
<br>
Anyone? I have hard time to believe this is not possible with
chan_pjsip.
<br>
<br>
Anyway, may I ask how people handle the following scenario which
I imagine should be quite common:
<br>
<br>
- I have internal extensions talk to each other using g722. so
their codec setting (with chan_sip now) is "allow=g722,ulaw"
<br>
- I have carriers trunks that handle ulaw only (allow=ulaw)
<br>
- calls between internal extensions naturally happen over g722
as its their preferred codec
<br>
- 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
<br>
<br>
Can somebody please advise how to achieve the same with
chan_pjsip?
<br>
<br>
Thanks,
<br>
<br>
Michael
<br>
<br>
</blockquote>
<br>
</blockquote>
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