Add close method to peer connection factory

Add close method to peer connection factory.

Supporting AWS Workspaces WebRTC Redirection

Adding AWS Workspaces WebRTC redirection support.

Note: Streams v2.14.4 and above is required for AWS Workspaces WebRTC redirection support.

Resolving Missed Call Issue for Citrix Integration

This release addresses the missed call issue for Citrix integration that could occur after long connection time, by upgrading the UCSDK dependency to v3.1.0

Fix missing standby peer connection issue

Fix missing standby peer connection issue which was introduced in v1.1.22

Supporting Citrix WebRTC Redirection

Adding Citrix WebRTC redirection support.
Note: Streams v2.12.0 and above is required for Citrix WebRTC redirection support.

Enforcing the usage of the standardized getStats API

In this release, we have made improvements to the getStats() function to ensure compatibility with Chrome's upcoming changes.

Background: RTCPeerConnection has two versions of getStats(), one that is spec-compliant returning the report via resolving a promise, and one that is non-standard returning a very different report via a callback as the first argument. The non-standard callback-based version of getStats() will be removed in future versions of Chrome.

To align with Chrome's future release, we have enforced the use of the standardized (promise-based) version of getStats() in this release. This change ensures a seamless transition by generating output that is consistent with the results obtained from the legacy getStats API. Please make sure to upgrade your connect-rtc-js to this version to support the Chrome future release. Please find more about the Chrome deprecation plan here: https://groups.google.com/a/chromium.org/g/blink-dev/c/PxQQtEM7za0/m/Dg-zX1bZBQAJ

Improved error handling for extracting MediaRtpStats

Improved error handling for getStats method. Motivation: #57

Supporting Wildcard in rtcp-fb

Important: Customers who directly integrate connect-rtc.js with their custom Amazon Connect softphone must upgrade to the 1.1.17+ version before April 3 2023, in order to support Google Chrome browser version 112+ and Microsoft Edge browser version 112+.

Release: Connect-RTC-JS now supports wildcard payload types in rtcp-fb SDP attribute lines. This is in line with Chrome’s plan to add rtcp-fb lines with wildcard payload types starting from version 112 (Announcement).

Allow user provided local stream

v1.1.16
1.1.16

Allows user provided media stream as local stream. This addresses Chrome unfocussed mic access issue. Applications(Streams etc.,) can grab mic access from focussed tab and provide that to RTC JS. RTC JS uses user provided local media stream to setup webRTC session.

Deprecating plan-b SDP

v1.1.14

1.1.14