[asterisk-users] rtp problem with 1.8.0-rdc1

[Philipp von Klitzing]

Hi!

>> Why is it a problem? It sounds like Asterisk does silence suppression.
> 
> 1) With no rtp traffic, the nat device will drop the connection in it's
> nat table and thus disconnecting the softphone from Asterisk. (after 
> the router's timeout period of course)
> 
> 2) The other issue is you are connected to a conference call and you 
> want to mute your transmitter while listening to the conference.

Set internaltiming to yes in asterisk.conf and see if that helps. In 
addition you might also be able to change the mute behaviour of your SIP 
clients so that it keeps on sending silent RTP packets.

Philipp

P.S.: You could also mute the conference user, not the SIP UA.