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-bounces at lists.digium.com [mailto:asterisk-users-bounces
at 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
at 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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