[Freeswitch-users] webrtc/Sip issue

[Michael Jerris]

a debug trace including the sip traffic might tell us more.

> On Jul 29, 2015, at 3:21 PM, Chris Mandra <mandra at gmail.com> wrote:
> 
> Hi guys - one of my users keeps getting this error in  fs_cli  when he tries to connect via Chrome (web-rtc) on a macbook running yosemite.
> 2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP REPORTS ERROR: [Remote Address Error!]
> 
> I'm using essentially the same setup and never have an issue. Any ideas? He doesn't get anything very useful in the chrome console, just this:
> 
> *********************
> 
> Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | closing INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m
> 
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.invitecontext.mediahandler | closing PeerConnection
> 
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog | dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted
> 
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event failed
> 
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event rejected
> 
> 
> sc-webrtc.js:931 _rejected called
> 
> *******************
> 
> 
> 
> Any ideas? We're using latest FS and sip.js
> 
> 
> 
> thanks,
> 
> chris
> 

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