[asterisk-users] Asterisk RTP Questions

[James Bensley]

Heh, should have guessed it would be you that replied Gareth ;)

Sorry yes, this box is on public IP with no NAT as is the upstream
providers box (or so they say).

So we have had audio cease outbound towards the provider. We have a
couple of volunteer customers who are being routed via this new test
upstream. It's very difficult (basically impossible!) to replicate the
failure it's so infrequent. Looking at PCAPs between us and the
upstream we stop sending them audio for example and then a little
while later the call drops. Without PCAPs between us and the customer
at the same time I can't say why we stopped sending audio (where we
receiving any from the customer, did their connection drop for
example).

We have also had the reverse where we stop receiving audio then a
short period later, SIP BYE from us to them!

I have read up on rtpkeepalive and rtptimeout. I will put this to one
side for now until we have a direct connect to the new test provider
there are to many variables in the equation.

Thanks for your input though Gareth!.

Kind regards,
James.