Conference Ports

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Hello again Norman,

I ran another test on two different PC's and did not get the disrupted
audio. It has to do with my "super fantastic enhanced" sound card I'm
sure. I used the same pjsua.exe in all test cases. Anyway thank you
for your help Nanang. If i succeed in tracing down the problem on my
PC I will let you know.

Best Regards,

Hubert

On Wed, Oct 8, 2008 at 8:52 PM, Nanang Izzuddin <nanang at pjsip.org> wrote:
> 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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>>
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>>
>
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