So, I've done some more testing and got some more info.

I have one endpoint that does silence suppression and one that doesn't. When 
the silence suppressing endpoint stops sending RTP, asterisk stops sending RTP 
to the other endpoint. I have disabled directmedia and directrtpsetup and it 
made no difference. I have even forced one endpoint to use GSM and the other to 
use ULAW (forcing asterisk to re encode everything) and asterisk STILL stops 
sending RTP when the endpoint does...


-----Original Message-----
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Tucker
Sent: Friday, 28 January 2011 11:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion 
(asterisk-users@lists.digium.com)'
Subject: [asterisk-users] RTP keepalive doesn't work

Hey guys,

I'm using asterisk 1.6.2.13 and have an endpoint which uses silence suppression 
which I can't turn off. I've set rtpkeepalive=10 in sip.conf [general], as well 
as under the peer details for our sip provider but it doesn't seem to do 
anything. Rtp debug shows that we are receiving RTP from the SIP provider, and 
forwarding it to the end point, but no RTP packets are sent back to the 
provider (ie. No keep alives).

I did find a bug report of this exact issue, but it was closed with the message 
to ask the mailing list...

Any ideas?

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