On 27/11/13 14:12, James Bensley wrote:
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)?
There isnt one really. There is a rtptimeout setting but that is
designed to hang up a call if no rtp has been received for X seconds.
Its going to normally be something long like 30 seconds as is designed
to end a call if the sip endpoint you were talking to dies.
How can I monitor for such an effect?
Does anyone else have any / or had any issue like this?
What direction does the audio stop?
have you looked at the SIP traces to see if there are and reinvites at
the same time?
You havent said if your server is directly connected to the internet
with its own public IP address or whether it goes via NAT.
Kind regards,
James.
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