Hi all, Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works.
My network setup by the way: I am working from a cable modem, I created the test setup at digital ocean. From my laptop I also have a direct VPN connection to the asterisk server my laptop being 192.168.241.10 and asterisk being 192.168.241.30 I think something is wrong with the RTP address negotiation, but I have trouble interpreting the SDP wrt WebRTC and ICE. 1. asterisk seems to be telling sipml5 to send audio to it's public ip addres, but * sends to 192.168.241.10 2. the asterisk output does show RTP flows to chrome, but there's no sound from chrome. I hope someone can intersperse the output with comments? Thanks, Antonio Asterisk console log, and Javascript console output: http://pastebin.com/dTFTrzg6
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