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.