bhavik patel
2014-May-10 07:27 UTC
[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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140510/84be2c63/attachment.html>
Rusty Newton
2014-May-14 16:28 UTC
[asterisk-users] Asterisk 11.9 with webRTC demo integration
On Sat, May 10, 2014 at 2:27 AM, bhavik patel <bhavikpatel14388 at gmail.com> wrote:> Hi All, ><snip>> 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>> 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
Olli Heiskanen
2014-May-14 16:59 UTC
[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 at gmail.com>:> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140514/4506a558/attachment.html>