I agree with Carlos, it's a bug, I have adapted the patch to FS v 1.4.20 and it is working, On Wed, Jul 15, 2015 at 6:01 PM, Carlos Ruiz Díaz <carlos.ruizdiaz@gmail.com
wrote:
There's a high change this being a bug in FS, caused by FF removing lots of cipher suites it used to use for the handshake.
I experienced this myself with rtpengine.
This is the commit that fixed the issue on that project, in case it may be useful for the maintainer(s) in charge of webRTC in FS. [1]
[1] https://github.com/sipwise/rtpengine/commit/21e1fb680762f421e05fab036f8138b2...
On Wed, Jul 15, 2015 at 10:47 AM, Igor Olhovskiy <igorolhovskiy@gmail.com> wrote:
I'm trying to figure out if it is a bug of FS :) So asking if someone is having the same )
2015-07-15 18:08 GMT+03:00 Brian West <brian@freeswitch.org>:
I would like to know what we can do to get the point across about filing bugs on JIRA?
Even if you file something and it ends up not being a bug, we can at the very least close it as 'not a bug'. What we can't do is mentally track every last issue thats reported on the list.
https://www.youtube.com/watch?v=lxB57TtX27I
Thanks,
On Wed, Jul 15, 2015 at 5:55 AM, Igor Olhovskiy <igorolhovskiy@gmail.com
wrote:
Hi!
I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to Firefox 39 is seems to be bad idea, cause call through it fails due to
2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake failure 1 2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL
On FF 38 and Chrome all is ok. Am I only one with this issue of it's really something wrong with FF?
-- Best regards, Igor
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-- Best regards, Igor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Carlos http://caruizdiaz.com
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