Routes: WebRTCAG Projects RTC Platforms (Routing Guides)
This guide describes the routing for WebRTC end-points.
The clients can be Web browsers or custom made applications. All must use the SylkRTC API provided by AG Projects.
Web clients use a WebSocket to connect to SylkServer components, which in term registers on behalf of client to the SIP Proxy/Registrar of the platform, thus emulating a regular SIP end-point to the SIP platform. Multiple web clients can connect using the same SIP account credentials, multiple web clients appear as separate SIP end-points registered on the SIP Proxy of the platform. Incoming calls addressed to a particular SIP account are forked to the web clients as well.
Supported media is peer-to-peer RTP audio/video and multi-party RTP audio/video.
Peer to Peer RTP audio/video media flows between the Web clients through SylkServer and MediaProxy. A Web client can call and receive calls from other SIP clients providing they use a compatible codec.
Web clients can create multi-party video conferences. These particular application is not using SIP protocol for signaling and is not traveling through the SIP core network.
The currently interoperable codecs between SIP and Web are Opus for audio and VP8/9 and H.264 for video.
Open using a web browser with WebRTC support to platform web gateway address. For example in case of SIP2SIP services open this url:
You can login in the web page with your SIP2SIP credentials.
Download Sylk Client from http://sylkserver.com
Install Sylk client for your particular desktop operating system.
Blink for iOS and Android are available for download from their respective app stores. The mobile clients are curently under development and may not provide all functions available in the desktop versions.