Revision for “Sipmobile solution for WebRTC” created on June 18, 2014 @ 11:52:08
Sipmobile solution for WebRTC
WebRTC is a new standard which brings voice and video communications to web browsers without additional client software and plugins. Currently supported browsers: Google Chrome (exempt iOS devices), Opera, FireFox. Advantages of WebRTC: 1. Platform and device independence. 2. No security risks with third party client software. 3. Secure communications - WebRTC always use encryption for voice and video. 4. High quality codecs - Opus for audio and VP8 for video. 5. Can be easily integrated with existing sites and business software (click to dial service for example). Currently there are some problems with integrating WebRTC clients and traditional SIP networks. Usually you will need some gateway solution to connect WebRTC clients to a SIP network. Sipmobile solution includes: 1. SIP Proxy which converts SIP transport from WebRTC to UDP, TCP or TLS, 2. Media gateway with automatic conversion of media streams between WebRTC and SIP clients. 3. Sample WebRTC Client - <a title="Sipmobile WebPhone" href="https://www.sipmobile.org/Webphone/" target="_blank">Sipmobile WebPhone</a>. 4. Android SIP client - <a title="Sipmobile softphone Android" href="https://www.sipmobile.org/sipmobile-softphone-android/" target="_blank">Sipmobile Softphone Android</a>, which is fully compatible with WebRTC client inside Sipmobile.org domain.