[Freeswitch-users] WebRTC + Chrome = RTP Socket Error!

[Brian West]

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 at 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 at 192.168.1.100 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>
> Has anyone experienced the same problem?
>
>
> --
> Luiz Fernando Softov
> http://www.softov.com.br
> fernando at softov.com.br
>
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-- 

*Brian West*
brian at freeswitch.org


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