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 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?
Thanks,
Michael
On 7/5/23 14:59, Michael Maier wrote:
Hello Michael,
you are referring to the following behavior - did I get it correctly?:
outbound broken: asterisk offers g722 / g711 to provider (callee),
callee answers g711. Asterisk now transcodes between caller and callee
(g722 <-> g711).
inbound works: call from provider: g711 -> asterisk drops g722 and
passes g711 to internal callee -> no transcoding.
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.
Thanks
Michael
On 05.07.23 at 17:58 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
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