I got a vm with asterisk in my local network, the configuration is as in https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5.
When making a call from PhonerLite to sipml client there is a delay (it also happens when asterisk before nat) (as shown on a screenshot 1)
Everything else works fine - rtp packets run, no call termination.
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<screenshot 1.png><screenshot 2.png>
On Feb 20, 2015, at 3:50 PM, Дмитрий Кутузов <dmit...@gmail.com> wrote:Is it really necessary to do ICE candidates gathering with only SIP calls with Asterisk server? What is the real purpose of ICE?Thanks