[Asterisk-Users] RTP problem

[trixter http://www.0xdecafbad.com]

On Thu, 2005-04-14 at 07:23 -0500, Eric Wieling wrote:

> Is Asterisk getting a stream of RTP packets from the SIP client?  What 
> happens if you start talking on the SIP device?  Does Asterisk then 
> start sending RTP?  It still sounds like VAD and silence supression is 
> enabled on the SIP device.
> 

Let me recap since it was spread over a couple emails back and forth
between someone and this is a high volume list.


Any sip clinet (local or remote) connecting to my asterisk system is met
with silence.  

All sip clients send a RTP stream correct

Asterisk reads DTMF from sip clients with no problem and executes the
appropriate dialplan entries

asterisk will hang on a playback() or background() call and will send
only 1 RTP packet to the SIP slient (note the sip client is sending RTP
the whole time, thus I do not think its silence detection on the client)

If I press a number that causes a different playback() asterisk will
send 1 more RTP packet and hang on that playback.

If I go through asterisk but the destination is a sip client the RTP
stream works perfectly.  The *only* time there is a RTP problem is when
the destination is asterisk.  musiconhold sends no rtp packets, playback
1 packet only.

SIP traffic goes back and forth, a sip show debug reveals proper call
setup/tear down.  

It does not matter if the sip client is local or remote.  

This setup was working a couple days ago, which makes me curious as to
what could possibly be happening.

-- 
Trixter http://www.0xdecafbad.com
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