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

Arnd Vehling av at nethead.de
Fri Sep 1 10:17:01 MST 2006


Joshua Colp wrote:
> Can you do an rtp debug while it is in this state and see what the 
> output is like? It would help immensely.

Dang. I just replaced the installation with an older version. I will
make a new install though. What exactly do you want?

The rtp boddy of the sip INVITEs and ACKS look good. The ip addresses
in the rtp body of the clients gets replaced with the ip address of
the asterisk server. I can give you a sip protocol trace (i use ngrep for
this) as soon as i have the new installation up but you need to tell me how 
you want the rtp stream captured/logged/send.

regards,

   Arnd




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