[Freeswitch-users] webrtc/Sip issue

[Chris Mandra]

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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