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.
Along with The OnSIP Network, OnSIP has also announced their contribution of an open source JavaScript library, SIP.js, to the web developer community. SIP.js allows WebRTC developers to add a necessary component, signaling, to facilitate in their applications peer-to-peer voice, video, messaging, and data channels. This new SIP stack is a fork of JsSIP and can be used with any SIP platform. When paired with The OnSIP Network, SIP.js enables developers to add a complete signaling layer to their applications without needing to build their own stack.
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.