[asterisk-users] RTP keepalive doesn't work

[Ryan Tucker]

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?