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[jwbensley at gmail.com]

Hi All,

I have some questions regarding RTP and Asterisk;

I am trialling a new SIP upstream provider. We connect to them over
the Internet at present which I know is not ideal, but we are just
testing at present. During the trials we have had an issue where we
have had one way audio between us and the provider after the call was
successfully set up and bidirectional audio has been already flowing
(so at some point during an existing call, two way audio has dropped
to one way audio).

I am running a constant PCAP which I sent back to the provider. They
have said that the latency has increased or fluctuated to the extent
that RTP as stopped sending audio in one direction (because of our
test peering over the Internet). Weather this is true or not is a
separate issue, what I want to know is;

What is the maximum delay RTP will tolerate one way (Does Asterisk
have a limit too)?

Can this be tuned (increased or decreased) within Asterisk (I'm
thinking of DSL customers where we may have this issue between our
PBXs and the customer)?

How can I monitor for such an effect?

Does anyone else have any / or had any issue like this?

Kind regards,
James.

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On 27/11/13 14:12, James Bensley wrote:

[jwbensley at gmail.com]

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.

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_____________________________________________________________________
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