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

Paul Belanger paul.belanger at polybeacon.com
Wed Jan 28 09:45:49 CST 2015

On Wed, Jan 28, 2015 at 8:27 AM, Antonio Gómez Soto
<antonio.gomez.soto at gmail.com> wrote:
> 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 and asterisk being
> 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
> 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?
Pastebin the fill debug, you've delete an important piece of information.

Paul Belanger | PolyBeacon, Inc.
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