OnSIP, a provider of business communications services, has announced the OnSIP Network
, a Platform as a Service that allows developers to add WebRTC to their web apps.
The OnSIP Network solves a similar problem for WebRTC over the Internet that phone numbers have for calling over Public Switched Telephone Network: How does one user find another user over the network? The OnSIP Network provides the ability for integrating immersive video, voice and messaging in any web application.
The platform provides the ability to establish real-time media sessions like voice and video calls using any codec of choice, handling the codec negotiation. It enables IM messaging between endpoints with SIP MESSAGE requests. Developers can establish a real-time data channel (RTCDataChannel) session between endpoints for file sharing, chat, gaming, etc.
The OnSIP Network also provides the SIP signaling controls needed to start a call, transfer a call, fork a call, end a call, etc. Each OnSIP user address can be registered on up to 10 endpoints at once (think simul-ring). Developers can manage their OnSIP account and registered endpoints in the OnSIP admin portal or can add account management to an app using the OnSIP Admin Service SDKs.
This open source project is now available on sipjs.com
, and developers who would like to start building
with The OnSIP Network can do so now.
Read more: http://www.onsip.com/webrtc-sip-network
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