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

Michael Jerris mike at jerris.com
Thu Feb 12 21:18:08 MSK 2015


I don't actually know the answer for that, but you could dig in the sofia-sip library in tport if you want to find the answer.  we hook into the tport logging for those.

> On Feb 12, 2015, at 11:50 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Thanks Brian, that's the plan.
> 
> What about my second question about logging of SIP traces?  Are they logged upon successful sending?  If so, is there a way to see in the logs when FS attempts to send?
> 
> Basically, I'd like to figure out where in the SIP process the delay happens.
> 
> On Thu, Feb 12, 2015 at 7:42 AM, Brian West <brian at freeswitch.org <mailto: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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150212/6a33ed00/attachment.html 


Join us at ClueCon 2016 Aug 8-12, 2016
More information about the FreeSWITCH-users mailing list