SIP.js 0.7.0 Adds Fixes and Support for Attended Transfer - Recommended Upgrade

We have just released a new version of SIP.js, our SIP Javascript stack that's perfect for developers who want to build WebRTC-based features. SIP.js 0.7.0 is now the most stable, comprehensive version of the library to date.

"SIP.js 0.7.0 is a minor version bump from 0.6.4," said OnSIP Developer Joseph Frazier. "It contains Attended Transfer support, improved Session and MediaHandler events, and a handful of other bug fixes. We recommend that all users upgrade to this latest version."

Here are some highlighted improvements:

• Our WebSocket dependency is now included using require('ws').  Thanks to Browserify, this shouldn't affect most users. Node.js users will have one less thing to worry about.
• We are now using Promises internally.  This has been the case on the Master branch for some time.  Node.js users will be polyfilled to use promiscuous, or Node.js native Promises if available.
• We are now using the Node.js native EventEmitter, bundled in Browserify.  There are slight changes to the EventEmitter API due to this change, but old methods are supported (though deprecated) for now.
• Session has a new terminated method.  (Okay, it's not new, but we fixed how it works and document it now.)
• Session termination events (rejected, failed, cancel, bye, and terminated) have been cleaned up and now behave more consistently, both in terms of internal behavior and RFC specs.
• The WebRTC MediaHandler now has many more events for ICE connection states and candidate gathering.
• Support for REFER with Replaces and INVITE with Replaces.
• A handful of other bug fixes.

So check it out! Grab SIP.js 0.7.0 today.