I realize you are now supporting WebRTC with SIP itself yet, but could
you comment on how Jitsi handles this DTLS-SRTP issue? Do you do things
the Asterisk way or the JsSIP/Chrome/Firefox way?
I put a hack into JSCommunicator to make it work with Asterisk from
Firefox but this may break wider compatibility and set a bad precedent
if Asterisk is not correct here.