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[asterisk-users] Asterisk 11.9 with webRTC demo integration

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bhavikpatel14388 at gm...
Guest

Posted: Sat May 10, 2014 2:28 am Post subject: [asterisk-users] Asterisk 11.9 with webRTC demo integration

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

rnewton at digium.com
Guest

Posted: Wed May 14, 2014 11:28 am Post subject: [asterisk-users] Asterisk 11.9 with webRTC demo integration

On Sat, May 10, 2014 at 2:27 AM, bhavik patel
<bhavikpatel14388@gmail.com> wrote:

Quote:

Hi All,

<snip>

Quote:

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".

That is a question about Chrome, not about Asterisk. A quick Google
search pulls up this information:

https://support.google.com/chrome/answer/2693767?hl=en

" If you select Allow on a "http" URL your preference will not be
remembered in future visits. If you select Allow on a "https" URL,
your preference will be remembered in future visits. "

<snip>

Quote:

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" .

Not sure what is going on here. You can try following my tutorial for
testing with the SIPML5 demo here:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
, It also uses Asterisk 11 and chan_sip which matches what you are
doing.

--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

--
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ohjelmistoarkkitehti a...
Guest

Posted: Wed May 14, 2014 11:59 am Post subject: [asterisk-users] Asterisk 11.9 with webRTC demo integration

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@gmail.com (bhavikpatel14388@gmail.com)>:

Quote:

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

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

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