[pjsip] Conference Ports

[Norman Franke]

I seem to recall having to generate silence in such cases. I wanted to  
implement mute, but ran into this problem. So I ended up just  
connecting it to a port of my own design that sent a stream of low- 
level random noise (as total silence sounded to the other end like the  
connection was terminated.)

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

On Oct 6, 2008, at 3:56 PM, Nanang Izzuddin 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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