SIP.js

Bug Fixes

• Fix potential out of bounds error with addMidLines modifier
• Do not send multiple CANCEL messages if cancel() is called multiple times (Fixes #618)

Modifiers

• Add modifier to add a=mid lines for FF63+
• Stop stripVideo modifier from creating circular object
• Make stripVideo modifier support removing video from bundles

Build

• Use typescript builder

Bug Fixes

• Better logging of SDH errors
• Add Transport closed event

Session Description Handler

• Automatically fallback to audio only call if video fails
• Make errors explicitly for SDH
• Errors do not keep falling through
• Deprecate GetDescriptionError in favor of SessionDescriptionHandlerError

Build

• Enable es6 in eslint
• Update webpack and remove uglifyjs-webpack

Bug Fixes

• Listen for onError event from Websocket
• Stop forcing lowercase for SIP URI parameters
• Fix ICE gathering complete on old browsers
• Transport: Accept multiple CRLFs as a keep alive packet
• Grammar: Accept 'IPv6reference' for Via received parameter

Bug fixes

• Fix transport constructor configuration check
• Fix keep alive debounce timeout
• SDH Fixes

Other changes

• Various test and build improvements
• Change npm to include unminified dist file

Bug fixes

• Fix this reference in acquire function of default session description handler
• Make invite return and do the bulk of logic async so that users have time to add event listeners before events get fired

Acquire media first for FF61+

• Add option for default session description handler
• Make the first GUM call for audio only

Simple

• Acquire media before setting description when using Firefox
• Add traceSip option back to Simple

Fixes

• Refactor SDH to work with FF61 undocumented change by adding local streams to PC before calling setRemoteDescription
• Merge pull request #573 from fgast/origin/fix-MaxForwards-in-CANCEL
• Introduce Modifiers.stripRtpPayload() to allow stripping of arbitrary
  RTP payloads. Updated existing modifiers to delegate to this new
  function. (Thanks @seanbright)
• Also a large number of small transport related fixes

Features

• WebSocket Transport decoupled from the UA
  • Transport-related UA config options have been moved to their own transportOptions config options, an object
• SIP.Transport now a generic interface
  • Allows creation of a custom transport layer to be supplied to the UA
• WebSocket Transport built off of new generic interface, is the default transport
• Changed WebRTC/ directory to Web/
• Merge Pull Request #551: support for outbound PUBLISH requests

Features

• Add in-band DTMF
• Add Session Description Handler Observer
• WebRTC SDH prefer addTrack over addStream

Bug Fixes

• Close SDH when session is cancelled
• emit terminated on renegotiationError
• Fix error handling around reinvite
• Fix bug processing NOTIFY message on REFER progress

Bug Fixes

• Fix regex in strip telephone event modifier
• Pass correct arguments to session description handler factory during early media construction
• Add mute/unmute events to Simple
• Add events for reinvite succeed failure

Be the target of Music On Hold (RFC 7088)

• Be target of MOH #479
• Various bug fixes in the Session Description Handler
• Add babel-loader to the builder

Refactor of REFER Handling.

• Pull Request #459
• API docs
• Transfer Guide

WebRTC Session Description Handler

• Various ICE checking code improvements
• Cleanup constraint code
• Add tests

WebRTC Modifiers

• Bug fixes on g722 removal modifier

WebRTC Simple

• Allow anonymous registration
• Fix hold/unhold not checking the state of the call correctly
• Fixed an instance of logger not being defined

UA

• Remove unused configuration options

Miscellaneous

• Change log messages on Timers to debug level
• Update various README files
• Remove test/repl.js that was no longer used

Bug Fixes

• Reinvite event added to session and hold/unhold events removed
• Fixed link in package.json
• Update dependencies
• Update main in package.json

• Bug fix for external ICE candidate generation
• Bug fix for session tracking

Fix compilation for minified SIP.js and add License to distributables

Session Description Handler replaces the media handler. See the new docs for more information. This is an API breaking change from 0.7.X.

Removed EventEmitter.off() use EventEmitter.removeListener()

New build process that does not use grunt. See BUILDING.md for more information.

New Simple interface.

Safari and new WebRTC api support.

Removal of hacks.

Initial removal of enviornment.

Lots of code cleanup and other bug fixes.

Various bug fixes for memory optimization and stopping SIP.js
Add more events
Add options to hold function
Add dist files to NPM release

Add rtcpMuxPolicy parameter on UA to support Asterisk users.

• Refactor SDP mangling and move it into WebRTC/MediaHandler
• Allow TURN servers with no credentials
• New method of setting Content-Type on SIP messages
• Various fixes for subscriptions
• Update grunt-contrib-jasmine to work on new versions of NodeJS
• Update and simplify helper to attach the media stream to DOM
• Update WS dependency to 1.0.1 and NodeJS minimum version to 0.12
• Remove Nodejs v5 testing from TravisCI
• Add hackStripTcp and hackCleanJitsiSdpImageattr
• README and THANKS file updates

Version 0.7.4 could not be installed on Windows. This should fix that. See #316 for details.