You have to load the module res_srtp (secure media) in Asterisk.
module load res_srtp.so
(this is a requirement to talk websocket)
If you don't have it, must build it and install it.
Cheers,
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==
Miguel Oyarzo
DevOps Engineer
I am using Asterisk 11.3.0 and just updated Nightly to 24.0a1 (2013-06017)
and get a SIP 488 Not Acceptable Here response.
I have no problems using the same Asterisk configuration and the same page
to make a call from Chrome.
I have seen other people post a similar issue, but I have not seen a
