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

[Michael Jerris]

Also, I recommend you move to use sip.js instead of jssip.  They are fairly similar but sip.js is more up to date and maintained.

> On Feb 12, 2015, at 10:42 AM, Brian West <brian at freeswitch.org> wrote:
> 
> 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 <mailto: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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