SIPML5 - SIP client for a browser based on WebRTC
It is finished!
WebRTC (Web Real Time Communication) is an extension of HTML5 for transferring voice and video directly from the browser, through "native" means.
Official site of the project - http://www.webrtc.org ( testing on Chrome )
')
And today, the young telecommunications company Doubango Telecom has announced the release of a free client for SIP communication, written in JavaScript / HTML5, sipML5, the source code of which is distributed under the terms of the GNU GPLv3 license and is available at code.google.com/p/sipml5
Supported features:
* Audio / Video call
* Instant messaging
* Presence
* Call Hold / Resume
* Call transfer
* Multi-line and multi-account support
* DTMF support using SIP INFO
To test the new feature, you need a Google Chrome browser with the included MediaStream and PeerConnection features, you can see the flags if you type in the address line chrome: // flags /
After activating the flags (for some reason I have everything in one), it is necessary to restart the browser.
You can try it yourself on the demo page (public account here - code.google.com/p/sipml5/wiki/Public_SIP_Servers , bob password for bob :-)
Call from Android phone:
Call from iPad:
In 2010, Chris Andersen (curated and founder of TED in the form in which it is) spoke on the topic “How WEB video drives global innovation” - Chris Anderson: for those who do not know English, roller has the ability to include translation in Russian subtitles - subtitles).
What does the emergence of SIP, built into the WEB with the help of SIPML5? This means removing the last barrier associated with the cumbersome and incompatibility of interactive voice and video applications for WEB.
What is not only "city" - ActiveX SIP softphones, Java applets, and what about Flash with its own protocol RTMP, which caused the appearance of various bridges between RTMP and SIP (for example, siprtmp ).
Now everything is very simple and in the hands of WEB developers.
It is believed that the emergence of PHP, on which even housewives could code WEB sites, was one of the important factors of the Internet boom.
It seems to me that the emergence of SIPML5 will cause a real boom of interactive voice and video WEB applications.
We have long had a WEB (Internet revolution), Asterisk (a revolution in telephony), it lacked only what would connect them together - the generally accepted and accessible WEB telephone client.
And so he appeared :-)
Come on, gentlemen! Include your imagination! Entering the market is now so affordable and simple as never before!
Here are the first swallows, which, well, very happy this news :-)
WebRTC (Web Real Time Communication) is an extension of HTML5 for transferring voice and video directly from the browser, through "native" means.
Official site of the project - http://www.webrtc.org ( testing on Chrome )
')
And today, the young telecommunications company Doubango Telecom has announced the release of a free client for SIP communication, written in JavaScript / HTML5, sipML5, the source code of which is distributed under the terms of the GNU GPLv3 license and is available at code.google.com/p/sipml5
Supported features:
* Audio / Video call
* Instant messaging
* Presence
* Call Hold / Resume
* Call transfer
* Multi-line and multi-account support
* DTMF support using SIP INFO
To test the new feature, you need a Google Chrome browser with the included MediaStream and PeerConnection features, you can see the flags if you type in the address line chrome: // flags /
After activating the flags (for some reason I have everything in one), it is necessary to restart the browser.
You can try it yourself on the demo page (public account here - code.google.com/p/sipml5/wiki/Public_SIP_Servers , bob password for bob :-)
Call from Android phone:
Call from iPad:
What does it mean?
In 2010, Chris Andersen (curated and founder of TED in the form in which it is) spoke on the topic “How WEB video drives global innovation” - Chris Anderson: for those who do not know English, roller has the ability to include translation in Russian subtitles - subtitles).
What does the emergence of SIP, built into the WEB with the help of SIPML5? This means removing the last barrier associated with the cumbersome and incompatibility of interactive voice and video applications for WEB.
What is not only "city" - ActiveX SIP softphones, Java applets, and what about Flash with its own protocol RTMP, which caused the appearance of various bridges between RTMP and SIP (for example, siprtmp ).
Now everything is very simple and in the hands of WEB developers.
It is believed that the emergence of PHP, on which even housewives could code WEB sites, was one of the important factors of the Internet boom.
It seems to me that the emergence of SIPML5 will cause a real boom of interactive voice and video WEB applications.
We have long had a WEB (Internet revolution), Asterisk (a revolution in telephony), it lacked only what would connect them together - the generally accepted and accessible WEB telephone client.
And so he appeared :-)
Come on, gentlemen! Include your imagination! Entering the market is now so affordable and simple as never before!
Here are the first swallows, which, well, very happy this news :-)
Source: https://habr.com/ru/post/144293/
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