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

Antonio Gómez Soto antonio.gomez.soto at gmail.com
Mon Jan 26 19:47:39 CST 2015


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
to the asterisk server my laptop being and asterisk being

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
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?


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

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