Conference Ports

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Hi Hubert,

That isn't normal actually. I've just tried the scenario using two
PJSUA instances on two separate PCs, unfortunately the distorted audio
problem didn't occur, everything seemed to be normal as expected (e.g:
one way audio after 'cd 0 1').

Did you do the test using PJSUA? If it is reproducible on PJSUA,
please share the PJSUA config. Also please send along log files & RTP
packets capture, in case it is not easily reproducible.

Regards,
nanang


On Wed, Oct 8, 2008 at 5:20 AM, Hubert Langevin
<hubertlangevin at gmail.com> wrote:
> Hi Nanang,
>
> Yes you are right it is not a new behavior of the conference bridge
> but i still have a problem even when i'm not talking to Asterisk. I
> setup two PJSIP user agents on two separate PCs and make a call
> between them. Everything works fine but as soon as i disconnect my
> sound card by doing "cd 0 1" the incoming audio becomes hugely
> distorted, on either user agents. Is this normal? Because port 1, the
> call i made, is still transmitting to my sound card.
>
> I'm a bit confused, do i always have to have my sound card
> transmitting to port 1 or should I do what Norman did, that is
> creating a new port that just transmit low-level noise?
>
> Best Regards,
>
> Hubert
>
>
> On Tue, Oct 7, 2008 at 5:56 AM, Nanang Izzuddin <nanang at pjsip.org> wrote:
>> Hi,
>>
>> AFAIK, there is no such 'new' behavior of the conference bridge. And I
>> just recalled I read this somewhere that (by default) Asterisk's clock
>> for sending RTP is triggered by receiving RTP. So if it doesn't
>> receive any RTP packets, it won't send any RTP packets too. This might
>> be the case.
>>
>> Regards,
>> nanang
>>
>>
>> On Mon, Oct 6, 2008 at 5:17 AM, Hubert Langevin
>> <hubertlangevin at gmail.com> wrote:
>>> Hi everyone,
>>>
>>> I just realised a "new" behavior from pjsip that i haven't seen
>>> before. I make a call from my user agent to a soft PBX, asterisk in
>>> this case, in order to get music on hold. When i make the call, the
>>> conference port for the new call gets created and i hear music on my
>>> headset. Eveything looks fine. My sound card sits on port 0 and the
>>> new call i made sits on port 1 and they are both transmitting to each
>>> other. However if i do "cd 0 1", that is stop my sound card to
>>> transmit to port 1, i can't hear the music anymore, even though i
>>> still have Port 1, the new conference port playing music on hold,
>>> transmitting to port 0.
>>>
>>> Any ideas?
>>>
>>> Hubert
>>>
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>>
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>>
>
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