I guessed it’s something to do with NAT – most likely at the softphone
end.
I’m using X-Lite in all softphone cases, so it’s literally the end user’s
PC and router that’s different. I tried STUN but that seemed to make things
worse where no audio was heard from both ends.
I’ll run some debug and check the remote SDP to see what it contains and go
from there.
One
way audio usually is a result of NAT.
Not all NAT routers operate in the same way, and not all softphones handle
it in the same way, so that may explain why it works in one location but not the
other. I assume the softphones were behind NAT?
Set the FS log level to debug, and check the IP in the remote SDP. You need
the phone's external IP in there. Phones usually determine that using STUN. Some
NAT routers have a feature called SIP ALG to rewrite it for phones not doing it
themselves, but that often will conflict with phones that have already put the
correct IP (and also cannot work with TLS).
In short check where RTP is being sent to is correct...
Without the correct IP FS also has a last-resort feature where it detects
the IP the caller sends audio from and replies to there (RTP auto-adjust). A log
entry indicates when that happens. But that requires FS to receive audio, which
means it can't work until the caller starts speaking after the call is answered
(so might miss hearing the callee say 'hello'), and the caller will never hear
ringing (unless generated locally by the phone when it gets the SIP 183/180
response).
Just to complicate things Amazon EC2 runs behind a NAT layer too AFAIK. I
don't have any personal experience with EC2.
-Steve
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