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

[Oleg Stolyar]

Thanks Michael!  I guess I'll have to if this keeps happening after the
upgrade to latest master.

On Thu, Feb 12, 2015 at 10:18 AM, Michael Jerris <mike at jerris.com> wrote:

> 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> 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>
>> 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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