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


 
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antonio.gomez.soto at ...
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PostPosted: Mon Jan 26, 2015 8:48 pm    Post subject: [asterisk-users] Need some help interpreting SDP on a failin Reply with quote

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
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