FAQ

Where to get help?

For any technical issue, please connect to our developer group: https://groups.google.com/group/doubango.

How to use sipML5 API in my project

The API is a single file and could be downloaded at http://code.google.com/p/sipml5/wiki/Downloads. To get started, please check the programmer's guide and API.

How to get notified for new updates?

To get notified, please subscribe to our mailing list at http://doubango.org/contact.html

Which operating systems are supported?

Any System supporting a web browser (Chrome, firefox, Safari or IE) built with http://en.wikipedia.org/wiki/WebRTC'>WebRTC features. We highly recommend using https://www.google.com/intl/en/chrome/browser/'>Google Chrome

Is it possible to call another HTML5 SIP client

Yes

I get "SIP/2.0 488 Not acceptable here" response when I try to make calls from sipml5.org to a softphone or PSTN. How can I fix this?

This means the remote endpoint doesn't support all WebRTC features. * Go to http://sipml5.org/expert.htm * Check "Enable RTCWeb Breaker" then click on "save" button * LogIn again and try to make calls... all should work as expected :)

Is it possible to call any SIP client?

Yes... but the client must support all standards defined by the http://tools.ietf.org/wg/rtcweb/'>RTCWeb working group. We recommend these clients: * http://code.google.com/p/boghe/'>Boghe 2.0.100.701 or later : Windows XP, Vista and 7 * http://code.google.com/p/imsdroid/'>IMSDroid 2.0.491 or later: Android * http://code.google.com/p/idoubs/'>iDoubs: iOS (iPad, iPhone, iPod Touch) and MAC OSX ...off course you don't need to install these applications if you are calling another HTML5 SIP client. For more information on how to configure these clients, please click here. We recommend checking the non-exhaustive list of Public SIP Servers known to work with sipML5.

Is it possible to use Asterisk 11 to make/receive calls?

More info at https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support Yes. Please check wiki page at http://code.google.com/p/sipml5/wiki/Asterisk Yes. You have to enable the RTCWeb Breaker on the expert window.

Is it possible to use FreeSWITCH to make/receive calls?

Yes. You have to enable the RTCWeb Breaker on the expert window.

Is it possible to run use sipml5 on Android and iOS?

Yes but it requires a browser with support for WebRTC and WebSocket. For now only Ericsson WebRTC implementation (https://labs.ericsson.com/blog/bowser-the-world-s-first-webrtc-enabled-mobile-browser) runs on these systems. Issues on Ericsson WebRTC implementation: https://groups.google.com/group/ericsson-labs-web-rtc/browse_thread/thread/4acadd0da3fce37c

How to reduce the size of the scripts before deploying

To compress the scripts (css and javascript) you have to execute: ./release.sh the compressed files will be generated inside release directory.