Debugging RTP packets sending

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In my situation i use FreeSWITCH.org that make by default RTP media 
relay by dynamically discovering if the peer is behind nat or not.
I also force FS to act as a proxy for the media.

Now something it's not working, FS does not receive RTP packets of one leg.

I need to find out what PJSIP is sending out to the FS (that is not 
receiving anything), also to understand if PJSIP is sending something 
and if it's sending something if the target IP/PORT is correct (on SDP 
attribute it seems correct).

Fabio

On 24/03/10 16.56, nir elkayam wrote:
> if it one way then this might be the solution. u can't usually have 
> one way connection when both sides are behaind symetric nat.
> this is because the public address of the other side is never reveled 
> when he doesn't sends any packets..
>
> On Wed, Mar 24, 2010 at 4:37 PM, Fabio Pietrosanti (naif) 
> <lists at infosecurity.ch <mailto:lists at infosecurity.ch>> wrote:
>
>     Hi all,
>
>     i am struggling with debugging a one-way audio situation.
>
>     I analyzed everything related to SIP and SDP attributes and
>     everything seems fine.
>
>     I look at PJSIP logs in realtime during debugging.
>
>     Particulary i am interested in:
>     - see every RTP packets sent out
>     - see to which address/port the RTP packets are sent out
>
>     Is there a way to check RTP related information by enabling some
>     macro and/or define?
>
>     I am already running with PJSIP at maximum debug level.
>
>     Fabio
>
>
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>
>
>
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>
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