[asterisk-users] Need some help interpreting SDP on a failing WebRTC connection

[Antonio Gómez Soto]

Hi,

I am trying to setup a WebRTC connection to asterisk 1.13.0.
Using Bria a regular SIP connection works, but using sipml5 on chrome, I
got nothing.

My network setup by the way: I am working behind a comcast cable modem, the
test setup is at digital ocean, and 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 do not understand several things:

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 shows one way RTP flow. There's no sound from chrome.

I am trying to debug, but need some explanation about the SDP with respect
to WebRTC and ICE,
I hope someone can intersperse the output with comments?

Thanks,
Antonio

Below are the asterisk log, and the Javascript console output.

http://pastebin.com/dTFTrzg6
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150126/e0d04214/attachment.html>