[Freeswitch-users] Strange intermittent issue with SIP signals over WebRTC

[Oleg Stolyar]

Hi guys,

Sorry for a long email.

I ran into this strange problem a couple of time over the last year where
my Freeswitch stops receiving or sending SIP signals from and to WebRTC
(JsSip) profile.

At the same time, SIP signals are coming through just fine on a different
profile which is regular SIP.

It lasted for about 15 min.

When the problem was over, I saw a bunch of BYE signals being sent to the
WebRTC users at about the same for all the calls that ended  during these
15 min.

Also, INVITEs from these users that were sent during the 15 minutes
actually got through (or at least showed in the logs) after the problem
cleared.

I have a fairly old FS - master from July 2014 but was wondering if anyone
else had this problem.

There are no errors in the logs and I cannot reproduce this at will.

One possibility is that the network connection is somehow holding up these
signals rather than FS.  So, I was wondering when during the process of
sending SIP signals over WebRTC does it get recorded in the log?  Is it
when the sending is successful or as soon as it's attempted?  This could
tell me whether FS doesn't even try to send the signal during this problem
period or whether it's trying but cannot get through to the network somehow.

Another piece of information - calls that were connected before the problem
started continued just fine and the media kept coming through.
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