Hello,

Yes. When I today understood to set rtcp_mux=yes, at least Chrome (60.0 beta) worked (quickly tested) as expected.

I'm sure that some day dtls_rekey can be set to the other value than 0 as well with Chrome.

Best regards,

Teijo

10.4.2017, 16.57, Matt Fredrickson kirjoitti:

On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkins <[email protected]> wrote:


On Fri, Apr 7, 2017 at 9:44 PM, Teijo <[email protected]> wrote:

Hello,

I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
problem until now which remained was that if dtls_rekey was set to the
value other than 0, call hanged up when using chrome after the time where
dtls_rekey was set.

I suppose that "bad media description" shown in Chrome's window which
causes call to fail, has appeared with Chromes newer versions (currently 58
beta installed) or with Asterisk 13.15.0. Audio codec I'm using is Opus.

Has somebody else encountered this problem, or more better resolved it?

Best regards,

Teijo

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Hi Teijo

Take a read of https://nimblea.pe/monkey-business/2017/01/19/webrtc-
asterisk-and-chrome-57/ :)


13.15.0 should address rtcp-mux issues.

If there are still issues outstanding, it might be worth reporting a bug on
issues.asterisk.org.

Best wishes :-)



Tämä osa viestin runkoa ladataan pyydettäessä.


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