All notable GA release changes to this project will be documented in this file.
The format is based on Keep a Changelog and this project adheres to Semantic Versioning.
Feature
- A new event named
onCallConnectedhas been introduced, which is triggered when the PSTN callee starts ringing.
Bug Fixes
- Removed unnecessary dependency.
Bug Fixes
- Renamed
DOMErrortoDOMExceptionin the underlying JSSIP library to support latest Typescript versions.
Bug Fixes
- Enhanced call handling functionality to support multiple executions of the
call()method.
Bug Fixes
- Improved error handling by emitting "LoginFailed" event upon unsuccessful creation of User Agent (UA).
- Added a check to prevent sending DTMF signals when there is no internet connection, which was causing call disconnections.
- Enhanced WebSocket connection optimization and improved fallback mechanisms to reduce delays in establishing a connection.
- Streamlined the process for reconnecting active calls during network changes for improved stability and reliability.
- Improved SDK reconnection logic to prevent redundant WebSocket connections and optimize login scenarios during active sessions.
- Implemented a fix for the graceful disconnection of calls when a network switch occurs while the call is in the ring state, addressing call quality issues
- Implemented an internet access check prior to registration.
- Limited the Logout() function to execute only during active sessions.
- Other minor bug fixes and improvements.
Feature
-
Enhanced the
callinfoobject by introducing new attributes: Reason, Protocol, ErrorCode, and Originator, providing more detailed statistics. *Implemented Plivo STUN Servers to enhance reliability. This modification is now customizable through the 'usePlivoStunServer' flag which defaults to false. -
The reason for disconnection/connection is now published with the onConnectionChange event.
-
Introduced helper methods (isRegistered, isConnecting, and isConnected) for checking the client connection status.
-
Introduced a new event 'remoteAudioStatus' that signifies the reception status of audio packets from the remote caller.
Note: This feature works only in case of conference and MPC
-
Introduced a noise suppression feature to enhance audio quality by eliminating unwanted background noise during active calls. This feature can be enabled or disabled using the
enableNoiseReductionflag, which is set to false by default. Additionally, a new event,onNoiseReductionReady, is triggered when the noise reduction is ready to commence.Note: This functionality is not compatible with Safari.
-
onMediaPermissionevent will be triggered when media permission is revoked. -
Users will receive a
mediaMetricevent when speaking while the SDK is muted.Note: This functionality is not compatible with Firefox.
Bug Fixes
- Improved error handling by emitting "LoginFailed" event upon unsuccessful creation of User Agent (UA).
- Implemented a check to prevent sending DTMF signals when there is no internet connection.
Bug Fixes
- Enhanced WebSocket connection optimization and refined fallback mechanisms to minimize delays in establishing a connection.
- Streamlined the process for reconnecting active calls during network changes for improved stability and reliability.
Feature
- Implemented webRTC logging service for monitoring calls.
Bug fix
- Resolved an issue where noise reduction failed to initiate as anticipated due to necessitating a user gesture.
Feature
- A new event named
onNoiseReductionReadyhas been introduced, which is triggered when the noise reduction is prepared to commence.
Bug Fix
- Resolved a race condition that led to unexpected results when emitting the 'onLoginFailed' event.
Features
- Implemented new local hangup reasons to provide additional insights into the cause of call hangups.
Bug Fix
- Graceful disconnection of the websocket before emitting
onLoginFailed. - Resolved a type error issue resulting in: "Cannot assign to read-only property 'uri' of object".
- Preventing the establishment of a new websocket connection if there is an ongoing connection.
Note: The errorCode in the callinfo object has been renamed to code.
Features
- Enhanced CallInfo Object by removing
remoteCancelReasonand introducing granular attributes:reason,protocol,errorCode, andoriginator. - Removed google STUN and implemented plivo STUN Servers to improve reliability. This change is now configurable via the
usePlivoStunServerflag.
Bug Fixes
- Resolved type error issue causing
getHeader is not a function. - Improved call management: Calls will now disconnect automatically in the event of a network switch while in the Ringing state.
Features
- Enhanced Heartbeat timer for improved network monitoring.
- The reason for disconnection/connection is published with the onConnectionChange event.
- Streamlined the gathering of local media stream during the ringing state for incoming calls.
- Introduced helper methods (isRegistered, isConnecting, and isConnected) for checking the client connection status.
Bug Fixes
- Implemented automatic restart of heartbeat service when the browser fails to reconnect the socket.
Features
- Introduced a new event 'remoteAudioStatus' that signifies the reception status of audio packets from the remote caller.
Note: This feature works only in case of conference and MPC
Bug Fixes
- Enhanced SDK reconnection logic to optimize scenarios where users attempt to log in while already logged in.
- Added internet access check before registration.
Bug Fixes
- Fixed an "Invalid State" error that occurred during call hang-up.
- Addressed a Type error encountered when configuring output media devices.
- Limited the Logout() function to execute only during active sessions.
Features
- Trigger an event when media permission is revoked.
- Notify users when they speak while muted.
Bug Fixes
- Fixed: Call mutes when input/output devices change.
Features
- Included noise suppression feature to enhance audio quality, which effectively eliminates unwanted background noise during active calls. You can enable or disable this feature using the
enableNoiseReductionflag.
Note: This functionality is not compatible with Safari.
Features
- Added a new property
remoteCancelReasonunder theCallInfoobject, which indicates the reason for call termination for both incoming and outgoing calls.
Note: remoteCancelReason will be none during call state: ringing, answered.
Bug Fixes
- Fixed: Removed support for the getStats API, as it is no longer available in Chrome versions 117 and beyond.
- Fixed: Removed the predetectOWA functionality.
- Fixed: Issue on audio input/output device mismatch on windows platform.
Features
- Added: A new
useDefaultAudioDeviceflag, which allows the SDK to use either the system's default audio device or the recently added device for both input and output on Windows platform.
Bug Fixes
- Fixed: Corrected the handling of stir-verification in incoming call headers.
- Fixed: Fixed audio level discrepancies that occurred when changing input/output devices, ensuring accurate device settings.
- Fixed: Removed DOMError to support latest Typescript versions.
- Fixed: Restored functionality for incoming calls with PCMU codec.
- Fixed: Prevented SDK from logging out when re-registration timed out.
- Fixed: Reduced the time for firing the onConnectionChange event with a disconnected state to within 10 seconds when the SDK disconnects from Plivo servers, previously occurring within a 2-minute interval.
Features
- Added: A
refreshRegistrationTimerflag for user-configurable periodic re-registration by the SDK. - Added: An
onDtmfReceivedevent triggered when the SDK receives DTMF tones. - Added: Enhanced remote debugging with the collection and transmission of logs to Plivo servers.
- Added: Plivo STUN servers to ensure stable connections.
- Added: A
CALL_RINGINGevent signaling the initiation of incoming/outgoing call ringing to Plivo.
Features
- Added: New Client Region South-Asia to the existing Client Region list.
- Fixed: The call summary stats are not pushed to Plivo when the call is answered and hung up before the stats socket is open
Features
- Added: JavaScript framework detection for enhanced debugging.
Features
- Added: New call quality metrics added in Browser SDK: googEchoCancellationReturnLoss googEchoCancellationReturnLossEnhancement googJitterBufferMs.
- Added: On network change, a new event, ‘CALL_NETWORK’, will now be sent to Call Insights.
Bug Fixes
- Fixed: Issue with audio input/output device toggle on Windows platform.
- Fixed: Issue where non-default behavior was not maintained when Bluetooth was added over Headphones in Electron
Features
- Added: Attribute called ‘callerName’ to the onIncoming call event. This attribute contains the name of the caller (if set by the initiator of the call) and can be displayed on the user interface.
- Added: Ability to identify custom modifications to the officially released SDK versions.
- Added: Capture audio input/output device toggle events during an active call
- Added: Ability to select between Inband and Outband DTMF during initialization. For more information, refer to the Configuration Parameters section in the detailed reference.
Bug Fixes
- Fixed: Issue where one-way audio was observed for the next call after a custom device for input and output was removed.
- Fixed: Issue where call-summary event was not getting sent reliably under unstable network connections.
- Fixed: Issue where an error message was being published (setremotedescriptionfailed) when an outgoing PSTN call is rejected by the destination.
- Fixed: Issue where one-way audio was observed if an external bluetooth device was first connected and then disconnected whilst the Browser SDK was on an active call.
- Fixed: An issue where under unstable network conditions, Browser SDK keeps sending media metrics events even after a call has been hung up from the other end.
- Fixed: Issue where audio was getting picked up from both the external and internal microphones when the Bluetooth device was disconnected during a call and then added back during the next call.
- Fixed: Issue where audio output did not flow through an external Bluetooth device if one was added to a Windows machine.
Bug Fixes
- Fixed: Missing Information in Call summary event when the browser is closed during an ongoing call.
- Fixed: Issue where, for the first call post initialization of the SDK, input audio was getting picked from the device microphone even if an external Bluetooth device was added before calling.
Features
- Added: TypeScript support for Plivo Browser SDK
- Added: Network change improvements for Plivo Browser SDK
- Added: Improvements in events that were getting emitted during mute and unmute
Bug Fixes
- Fixed: Issue where incoming calls were not ringing in desktop browsers as well when the tab with the call was in background. This is the expected behavior in mobile browsers but not desktop ones.
- Fixed: Issue where an ongoing browser call would face a one-way audio issue (remote user won't be able to hear the browser app user) if the bluetooth audio device being used by the browser app user got disconnected and switched to another audio input device.
Bug Fixes
- Fixed: Issue where client name was being sent as 'Chrome' as part of Call Answered and Call Summary events even for calls made from Microsoft Edge browser. Now the value sent will be 'Edge'.
- Fixed: Issue where setting the configuration parameter 'dscp' as true was not behaving as intended and all UDP packets were still being set with DSCP Class CS0 (default class) instead of DSCP class EF (Expedited Forwarding) which is the expected behavior. Being set with EF ensures packets being tagged as high priority by network routers leading to lower chances of said packets being dropped and minimum per-hop delays.
- Fixed: Issue where, on receiving an incoming mobile browser call with the browser in background, the phone would start ringing despite a visual notification not possible (due to limitations imposed from browsers) and the customer having no way to find out which app is making the phone ring. Now the phone won't ring either.
- Fixed: Issue where a "TypeError" was being emitted if audio devices were toggled during an ongoing call. While this error did not affect any functionality, it was unnecessary and hence will not be emitted any more.
Bug Fixes
- Fixed: Issue where onConnectionChange event was not getting triggered in the following scenarios: normal endpoint logout abrupt socket disconnection due to changed network conditions
Features
- Added: New flag to selectively enable local or remote call quality tracking for Browser SDK. The old flag, which could only toggle both of these together, is to be deprecated in the next major release
##v2.1.33 (released@ 17-09-2020) Bug Fixes
- Fixed: Issue where incorrect audio level was getting printed if audio input device (mic) is switched from one input source to another (Eg: from bluetooth mic to internal mic) during an ongoing call.
- Fixed: Issue where one-way audio was being observed on calls made from Safari browser if audio input device (mic) is switched from one input source to another (Eg: from bluetooth mic to internal mic) during an ongoing call.
- Fixed: Issue where playback tone (incoming call ringtone) was getting played on device speakers if keyboard / headphones' 'play' button is pressed with the device in idle state.
Features
- Added: Buffer mechanism for temporarily storing Call Insights stats in the Browser before being sent to Plivo server in case of unstable network conditions.
##v2.1.32 (released@ 27-08-2020) Bug Fixes
- Fixed: Issue where Call Summary and Call Answered events were not getting sent if the user adds "sip:" in their username
- Fixed: Issue related to device "set()" API not working as expected when called during idle state (no active calls)
Features
- Added: Basic Call-insights RTP stats support for Safari
- Added: Active Input Device Info as part of Call Answered and Call Summary events for Safari
Bug Fixes
- Fixed: onMediaConnected event not emitting
Bug Fixes
- Fixed: Unwanted warnings that were getting printed while loading Browser SDKs.
- Fixed: 'Mute' state from a call was being retained to the next call.
- Fixed: Error that was being generated while trying to play a ringtone in Safari
Feature
- Added: Improvements in the audio level calculation for Call-Insights
Fix
- Fix for Heartbeat to stats-socket when the endpoint is logged out.
- Fix for Fractional loss during 'nan' which affecting the other stats
- Call-summary not getting sent when the browser is closed abruptly.
Fix
- Fix for handling only anonymise local IPs(mDNS) during media negotiation
Fix
- mDNS improvements
Fix
- Rtpstats improvements in Firefox Browser
Fix
- Removed callstats.io account
Fix
- Fixed mute/unmute behaviour during audio device toggling for ongoing call.
- Fixed the issue where microphone was still being accessed in the idle state(no calls)
- "onVolume" fix : It is now supported for external audio devices that get connected during the call
Feature
- Configurable maxAverageBitrate for OPUS codec
Fix
- Fixed issue related to call summary event
Feature
- Added Close protection flag for Browser
- Active device info to CallInsights
- Added Volume Indicator for Local and Remote audio
Feature
- Automatic input device (mic) fallback to a working input device while on call.
Fix
- Ability to change input devices while on call.
- Updated extra header handling.
- Added SDK version in feedback stats.
Fix
- Chrome's unified plan.
- Updates in MoS score calculation.
- RTP enabled flag in media metrics.
- Hangup issues in firefox and safari.
- Added stream for media metrics.
Fix
- Calculate fraction loss for 0-5 seconds.
- Send setup options in call answered and summary stats.
- Reconnect media when there is a network change.
Fix
- Disable RTP connection timeout flag fix.
Fix
- Chromium os is identified and sent to call insights.
- Showing appropriate error message for https only support.
Feature
- Added audio device info and media connection info to call insights.
Bug
- Changed the log from error to info
Fix
- Implemented onMediaConnected event when media connection established.
Fix
- Handled get stats in latest firefox version.
Fix
- Handled null values in stats calculation
- Added logs for callstats.io
Bug
- Added log level hierarchy for the logger.
Bug
- CallStats WebSocket was not connected due to callstats key not available when trying to connect
Fix
- Removed username and password from request params
Fix
- Made sendConsoleLogs parameter optional in submitCallQualityFeedback API
Bugs
- Added callinsights support for Firefox version 60 and above and Chrome version 64 and above.
- New Feedback API for customers to report an issue and choose to allow us to collect call related logs.
- Fixed fraction loss calculation for local stream
- Updated mos score by taking minimum value of local and remote mos.
Bugs
- Made plan-b default for upcoming chrome change in version 72.
Bugs
- Fixed key names camel case
Feature
- Extra meta data like browser's version and network informations for call insights.
Bugs
- Fixed a scenario where ongoing call audio getting paused when incoming call is rejected in mutiple incoming call scenario
- Added validation for reject and ignore functions in multiple incoming calls
Bugs
- Fixed a scenario where the calls were disconnection in chrome 54 due to the use of new webrtc APIs
Feature
- An extra option "actionOnOtherIncomingCalls" is added for answer(callUUID, actionOnOtherIncomingCalls) function. During a mutiple incoming calls scenario, if "letring" is passed for actionOnOtherIncomingCalls, the call with callUUID passed to answer function will be answered and other calls will be ringing silently. If no values are passed for actionOnOtherIncomingCalls, the other incoming calls will be rejected.
BUG fix
- microphoneDevices.set was setting deviceIds without removing the old deviceId, which is fixed.
- Added extra header to the callInfo object which is sent to some event callbacks
- Login with the endpoint which is currently logined in will not be allowed.
- Login with an different enpoint during an ongoing call will throw error saying "Cannot login when there is an ongoing call". [CSDK-87]
- Workaround for chrome bug where incoming call ringtone file was not loading properly sometimes which leads to incoming call without ringtone [SUP-272].
- Removed media metrics' dependency with callstats.io and used our call insights data [CSDK-109].
- If the call uuid passed in the answer function deos not match any of the incoming calls, a error message will be logged and false will be returned.
- Workaround for firefox bug where 180's SDP during outbound call should have a=mid line.
Feature
- An extra option "actionOnOtherIncomingCalls" is added for answer(callUUID, actionOnOtherIncomingCalls) function. During a mutiple incoming calls scenario, if "ignore" is passed for actionOnOtherIncomingCalls, the call with callUUID passed to answer function will be answered and other calls will be ignored. If no values are passed for actionOnOtherIncomingCalls, the other incoming calls will be rejected.
- New option to allow multiple incoming calls.
- New method
ignore()to take action on the incoming call. - Call insights data are collected for the insights enabled accounts.
- Made project publishable to npmjs ->
npm install plivo-browser-sdk --save - getPeerConnection() will return RTCPeerConnection object even when the outbound call is in ringing state.
- Added support for '-' in extra headers keys.
- Workaround for firefox bug where 180's SDP during outbound call should have a=mid line.
- Workaround for chrome bug where incoming call ringtone file was not loading properly sometimes which leads to incoming call without ringtone.
- Login with the endpoint which is currently logined in will not be allowed.
- Login with an different enpoint during an ongoing call will throw error saying "Cannot login when there is an ongoing call".
-
JsSIP v3.2.11 upgrade bug fixes
-
reduced ice gathering timeout to 2 secs
-
removed dependency with '_is_confirmed' variable
-
-
Callstats.io version upgraded to v3.53.1
-
Switched off pre-call-test of callstats.io
-
endpoint registration status fix
BUG fix
Fixed: Early Media playback on Firefox
Added
- Feature: WebSocket Connection change event listener . Detects abrupt websocket disconnection / connection / reconnection and notifies once per change.
- Upgrade Better logging for exception and unexpected behavior
- Optimize and upgrade npm dependencies such as Gulp and associated modules
- Upgrade underlying JSSIP library for ES6 , NOTIFY , REFER , INFO , RTCSessionDescription, Registrar
- Don't use pranswer for early media. Instead create an answer and do a workaround when the 200 arrives.
- Fix UA's disconnect event by properly providing an object with all the documente fields
- Add registrationExpiring event
- Don't send a Register request if another is on progress.
- RTCSession: process INFO in early state.
- Dialog: ACK to initial INVITE could have lower CSeq than current remote_cseq.
- WebSocketInterface: Add 'via_transport' setter.
- Use promise chaining to prevent PeerConnection state race conditions.
- New UA configuration parameter 'session_timers_refresh_method'.
- DigestAuthentication: fix 'auth-int' qop authentication
- RTCSession: emit 'sdp' event before creating offer/answer etc
- Unit test cases with linphonec
deprecated
BUG fix
Optimize local storage values
Added
- Feature: WebSocket Connection change event listener . Detects abrupt websocket disconnection and notifies one time per 30 seconds .
- GDPR upgrade in Callstats
deprecated
BUG fix
- Extra header length increased to 120, earlier it was 48
Added
- Add extra custom header
deprecated
BUG fix
- Fixed: Twilio webrtc API gets overridden by Plivo Sdk, Don’t alter URL.createObjectURL native code.
Added
- Feature: a config param
preDetectOwawithtrue/false, Detect one way audio before answering/sending the call. Default valuefalse - Feature:
audioDeviceChangeevent to listen for USB audio device changes. This event will emit an object with two propertieschangeanddevice.change- "added" or "removed"device- device specific properties - Collect Application logs in callstats dashboard under "logs" menu. Call summary log will get added to each callUUID.
- Callstats lib updated to 3.19.12, which gives callback based getStats once again in chrome 58
deprecated
BUG fix
Added
deprecated
- Removed: predetect OWA is taking 15sec in case of double natted system Refer: 894bcac0-1fc4-11e7-8451-8dbf96fbabce
BUG fix
- Packetloss was not emitted properly. values will be in decimals. Multiply by 100 to convert to %, Eg: packet loss of 2% will be emitted in value as 0.02
Added
- Feature added:
clientRegionproperty in initialisation options to set and route calls to specific MediaServer POPs. Allowed regions are["usa_west","usa_east","australia","europe","asia","south_america"]. - Feature added: Pre-detect One way audio. Before accepting Inbound call and before making an Outbound call. Make local peerConnection in loop and check for mic issues. This happens every first call on browser reload and then in 1 hr interval.
- Feature added: Call Terminated by caller, Callee details.
onCallTerminatedevent will have an object{'originator':'local'}if caller ends or{'originator':'remote'}if receiver ends - Feature added:
sendQualityFeedback()will now allow custom comments with a cap of 200 characters max. - Feature added:
debug:"ALL-PLAIN"in Options to turn off colour mode debug: "DEBUG" will show all logs except SIP trace debug: "ALL" will show all logs including SIP trace , Colour mode ON debug: "ALL-PLAIN" will show all logs including SIP trace, Colour mode OFF
deprecated
BUG fix
- on logout() - use stop() instead of unregister(‘all’);
- URL.createObjectURL(stream) is deprecated! Use elem.srcObject = stream instead!
- reject () - only if call is not answered.
Patch in JsSIP
- @line:1538 patch included, The moment we get one Public IP from ICE just send out INVITE. File path sipLib/RTCSession.js
Added
- Feature added: Even if users don’t set enableTracking in options, we should set enableTracking=true
- Feature added: mediaMetrics Alert if ICE gathering takes more than 2 sec either for outgoing call invite or incoming call answer. Event name
ice_timeout - Feature added: setConnectTone(true), Dial beep will play till we get 18X response from server. setting
falsewill not play beep tone.
deprecated
BUG fix
- Terminate ICE gathering in 2 sec. After upgrading to JsSIP 3.0.0 this
gatheringTimeoutwas removed.
Patch in JsSIP
- @line:1552 patch included to handle
gatheringTimeoutin sipLib/RTCSession.js; report to JsSIP versatica/JsSIP#432
Added
- Included JsSIP lib as
sipLibinsideplivo-websdk-2.0to handle customization in Jssip library
deprecated
BUG fix
- Initialise JsSIP only after checking for DEBUG in log level to show proper SIP trace
Added
- Play remoteStream if Incoming 183 has SDP
- Added callUUID to both incoming and outgoing calls in logs. Makes easy to get callUUID directly from logs.
- Better clarity logs to both Incoming and Outgoing calls at each level
- Added log to show if Plivo sdk is initialised twice.
- Moved onIncomingCall event to emit on Incoming call progress. Previously its was emitted immediately after newRtcSession
- Now CallStats dashboard should Plivo websdk version in context, Under 'General' menu
- Microseconds is added to logger date
deprecated
BUG fix
- Emit
webrtcNotSupportedonly on document ready. - handle when callstats lib is not loaded
Added
- moved all s3 links like audio and callstats lib to CDN links. cloudFront as Primary and CDN77 as secondary
- Don't initialise plivowebSDK when callstats lib is not loaded
deprecated
BUG fix
Added
- Audio API to control Input and Output devices
- availableDevices to show all all available audio devices
- revealAudioDevices to force allow permission and list available devices
- microphoneDevices to set and use particular microphone device as input
- speakerDevices to set and use particuarl speaker device for dtmf, remote audio
- ringtoneDevices to set and use particular speaker device for incoming ringtone
deprecated
BUG fix
Added
- we used webRTC adapter , a shim to insulate apps from spec changes and prefix differences which can work across most browsers
- supports Firefox, but mediaMetrics is not available since firefox doesn't supports it
- supports Opera ( not fully tested)
- added 2 new methods getLastCallUUID, webRTC and a variable version
- dscp param in options to support QoS
- websocket min try to 2 and max retry to 20 in case client disconnects from socket server
- reduced stun servers to 2 to reduce the size of SDP and to reduce stun gathering time
- mediaMetrics supported in chrome and opera a major feature to trigger warning events during bad network and audio conditions
- upgraded Plivo websdk to latest jsSIP 3.0.0
deprecated