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antonio.gomez.soto at ... Guest
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Posted: Wed Jan 28, 2015 8:27 am Post subject: [asterisk-users] Cannot get my first WebRTC experiment to wo |
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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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paul.belanger at polyb... Guest
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Posted: Wed Jan 28, 2015 10:46 am Post subject: [asterisk-users] Cannot get my first WebRTC experiment to wo |
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On Wed, Jan 28, 2015 at 8:27 AM, Antonio Gómez Soto
<antonio.gomez.soto@gmail.com> wrote:
Quote: | 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: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
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