Antonio Gómez Soto
2015-Jan-27 01:47 UTC
[asterisk-users] Need some help interpreting SDP on a failing WebRTC connection
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150126/e0d04214/attachment.html>