[asterisk-dev] Anyone got "SIP/RTP" Working reliably using svn trunk?

[Raphael Jacquot]

Arnd Vehling wrote:
> Hi,
> 
> i am having severe problems with asterisk svn trunk. SIP/RTP is pretty 
> unreliable. Calls between 2 phones connected directly (sip) to the box 
> always fail to establish a correct rtp stream. Looks like an NAT issue 
> because the rtp stream failing/not getting setup is the one to the phone 
> behind a NAT box. NAT is setup correct though. Works with older asterisk 
> version.

better yet, I get a nice crasher as soon as I try to call out via SIP 
(see bug http://bugs.digium.com/view.php?id=7854 )