Re: Asterisk 11.9 with webRTC demo integration

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Hello,

I'm far from being an expert, but as far as I know when you use https in your website the browser will ask to use the audio devices only once and then remembers your decision. When using http it will ask every time. 

Sorry I can't be of more help but hope this helps.

cheers,
Olli


2014-05-10 10:27 GMT+03:00 bhavik patel <bhavikpatel14388@xxxxxxxxx>:

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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