[asterisk-users] Cannot get my first WebRTC experiment to work.

[Antonio Gómez Soto]

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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