On Wed, Jan 28, 2015 at 8:27 AM, Antonio Gómez Soto <[email protected]> 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 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? > Pastebin the fill debug, you've delete an important piece of information.
-- Paul Belanger | PolyBeacon, Inc. Jabber: [email protected] | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
