El 11/06/2014 1:52 p. m., Matthew Jordan escribió:
On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington <[email protected]
<mailto:[email protected]>> wrote:
Chrome 35 broke all of this.... you need to be using DTLS now I
believe.
I had working secure web sockets with asterisk 12.2.x and chrome
34.... and then google broke eveything :)
I have not yet got around to test out DTLS etc. with chrome 35
Just so I don't waste too much time when I go to test, does anyone
know if all that's required for DTLS on the asterisk side is the
following in sip.conf?
dtlsenable=yes
dtlsverify=yes
dtlsrekey=60
dtlscafile=/usr/local/share/ca-certificates/myCA.crt
dtlscertfile=/etc/ssl/mycert.com.pem
dtlssetup=actpass
I assume I also need TLS configs in http.conf
Signalling is independent of the media; DTLS only affects the media.
However, there are known issues with Chrome's negotiation of DTLS and
Asterisk - see https://issues.asterisk.org/jira/browse/ASTERISK-22961
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
It is broken in Chrome (firefox never had SDES) because the WebRTC
standard favoured the DTLS SRTP implementation instead of the SDES one.
The thing is that although Asterisk supports DTLS implementation, it
only supports SHA-1 hashing but both Firefox and Chrome work with
SHA-256. The patch proposed in ASTERISK-22961 is an effort to solve this
issue.
Best regards
-----------------------------------------------------------------------------------
Este mensaje y sus anexos son para uso exclusivo de sus destinatarios y puede
contener informacion confidencial y/o privada protegida legalmente. Si usted
no es el destinatario, se le notifica que cualquier distribucion o reproduccion
de este mensaje, o de cualquiera de sus anexos, esta estrictamente prohibida.
Si usted ha recibido este mensaje por error, por favor notifiquenos inmediatamente
y elimine su texto original, incluidos los anexos y destruya cualquier
reproduccion
del mismo. Las opiniones expresadas en este mensaje son responsabilidad
exclusiva
de quien las emite y no necesariamente reflejan la posicion de Millenium Phone
Center S.A, ni comprometen la responsabilidad institucional por el uso que el
destinatario haga de las mismas.
-------------------------------------------------------------------------------------
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
