Do you happen to have the full sip trace with debug log from FreeSWITCH?

sofia global siptrace on 



On Tue, Apr 7, 2015 at 5:00 AM, Luiz Fernando Softov <fernando@softov.com.br> wrote:
I'm testing WebRTC + Chrome

Using 

http://sipjs.com

When im using FireFox, its work fine, i can make and receive calls with audio and/or video

But when i try to use chrome.

The other extension rings, but when i answer the call 

I receive this error, and call hangup

2015-04-07 05:51:39.016628 [ERR] switch_core_media.c:5775 AUDIO RTP REPORTS ERROR: [Socket Error!]
2015-04-07 05:51:39.016628 [NOTICE] switch_core_media.c:5776 Hangup sofia/internal/1001@192.168.1.100 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]

Has anyone experienced the same problem?

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