[asterisk-users] Asterisk 11.9 with webRTC demo integration

[bhavik patel]

Hi All,

I am trying to configure webRTC phone example for SIPml5 and i found this
info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
.

I have asterisk 11.9.0 installed and downloaded source of SIPml5 from
http://code.google.com/p/sipml5/source/checkout I copied sample code into
web root directory and example loaded successfully and also able to
register 2 extensions.

I have tried both browser Google Chrome and Firefox with their latest
versions.

For asterisk, I made some configuration like below. Please check :
http://pastebin.com/7KCvtcNf

For Outbound calls : when i am dialling 8002 -> 8001 every time Chrome
Browser asking for allow microphone. Is there any way to disable asking
permission and allowing it by default ? when i allow microphone then SIpml5
phone showing like "Not Allow".

Here is the asterisk logs : http://pastebin.com/JZeDjyay

For Incoming calls : When call come to browser,And allow microphone then
Call rejected and asterisk showing like "Got SIP response 603 "Failed to
get local SDP" in asterisk CLI.

But After some google i found new link
https://code.google.com/p/sipml5/wiki/Downloads for "SIPml-api.js" and
after replacing that JS File Calls are comming in browser even i am able to
answer that calls,Also in browser it says "In call" but in asterisk CLI it
keep showing ringing and other end showing like "remote ringing" .

Here is the asterisk logs : http://pastebin.com/e8Ap3bhq

Can anyone please let me know what am i doing wrong?


-- 
Thanks,
Bhavik Patel
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