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

[Brian West]

Update your FreeSWITCH, If you wish to use WebRTC you MUST use master, or
at least something more recent than last July. :P

On Thu, Feb 12, 2015 at 4:08 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> 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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-- 

*Brian West*
brian at freeswitch.org


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