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

[Joshua Colp]

Arnd Vehling wrote:
> 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
> 

You just need to type rtp debug in the Asterisk CLI, that will tell me 
all I need to know. Also - are you doing reinvites at all?

Joshua Colp
Digium