[asterisk-users] RTP keepalive doesn't work

[Ryan Tucker]

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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