Real use of WebRTC in IP telephony services

On Habré it was already reported about technology WebRTC (Web Real-Time Communications). The WebRTC standard, which is currently still in draft form, is predicting a great future, and in the future, the development of the Internet and web technologies is quite ready to follow the Skype path.
WebRTC can be used to create a fundamentally new category of web applications that can work with voice and video traffic without involving third-party technologies and programs using only HTML5 and JavaScript, which will greatly facilitate the development of applications for developers.

The technology is truly cross-platform, independent of the operating system you are using, you only need WebRTC support in your browser. Now it is supported by the popular Google Crome browser without any additional settings and plug-in installations on all desktop operating systems, such as Windows, Linux, Mac OS X and others. Also, due to the fact that for web calls, and in particular to establish a connection via SIP, html5 transport is required in the form of WebSocket technology , WebRTC can be used on almost all other Windows browsers (Safari, FireFox, IE, Opera ) by installing the optional webrtc4all extension .

The first service that launched commercial calls from a web browser was the serviceCallbacker , who integrated a web phone into his personal account, based on the open source sipml5 product .

What was done:
  • integration into sipml5 personal account and its customization with cancellation of additional functions that do not work or work, but are unstable, in the experimental testing mode;
  • installation and configuration of patched Asterisk 11 revision 373330. Asterisk in the latest release 11 supports WebSocket and SAVPF transport, which makes it compatible with the sipml5 web phone. SRTP support is required, so we build Asterisk with the required parameters: --with-crypto --with-ssl --with-srtp. Configuring Asterisk is no different from the usual one, with the exception of the http settings for WebSocket.

In the Asterisk configuration files:


avpf=yes ;необходимый параметр для работы веб-телефона, включающий
AVPF-транспорт для медиа-потока
encryption=yes ;шифрование SRTP
transport=udp,ws,wss ;ws и wss - транспорт для WebSocket

In fact, the user’s encryption flag can be set as encryption = no, otherwise other softphones or VoIP devices will also need to work with encryption, which is not always supported and may be inconvenient.

In order to try how it all works, you must have a login and password for the account of one of the Callbacker programs for iOS or Android, or you can register again and test the connection. Here is a personal account , in which there is the same web-phone. Currently, the Callbacker project is developing, gaining new opportunities. It is encouraging that the latest promising technologies, including those in the field of Internet telephony, do not die on the disks of SVN hosts, but find practical application.

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