0.11.6

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)

0.11.5

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

0.11.4

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

0.11.3

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

0.11.2

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

0.11.1

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

0.11.0

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

0.10.0

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

0.9.2

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

0.9.1

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