telnyx_webrtc 2.0.0
telnyx_webrtc: ^2.0.0 copied to clipboard
Enable real-time communication with WebRTC and Telnyx. Create and receive calls on Android, iOS and Web platforms
2.0.0 - 2025-06-13 #
Enhancement #
- Enhanced call state reporting to include more detailed information about the call state changes.
- Enhanced error reporting to provide more context on errors encountered during call handling.
- Simplified push notification decline process.
Bug Fixing #
- Fixed an issue where, on the Android and iOS clients, we weren't checking if the socket was open before sending messages. This could lead to 'StreamSink is closed' errors in niche edge cases.
1.2.0 - 2025-05-12 #
Enhancement #
- Added WebRTC Call Quality Metrics for each initiated call (enabled via debug bool on invite or accept)
- Adjusted logging to be more clear for connection process and websocket messages
1.1.3 - 2025-04-24 #
Bug Fixing #
- Clear Push Metadata in more places to prevent issue where a call is attempting to be attached based on previously stored push metadata. You can also now manually clear the push metadata by calling
clearPushMetadatamethod on theTelnyxViewModelclass. - Bump negotiation timeout to from 300ms to 500ms (per ice candidate)
1.1.2 - 2025-04-09 #
Bug Fixing #
- CodecError fix. Enhanced Ice Candidate Collection when answering calls to ensure more suitable candidates are used in the SDP.
1.1.0 - 2025-02-26 #
Enhancement #
- Added method to disable push notifications on the SDK via
disablePushNotificationsmethod. - Added a new parameter to the login configuration that allows users to provide their own
CustomLoggerto the SDK. This is useful for users that want to log the SDK's output in their own way or redirect logs to a server. - Added a new CallStates 'dropped' and 'reconnecting' to the
CallStateenum. This will allow users to know when a call has been dropped or is in the process of reconnecting. There will be aNetworkReasonprovided for both of these states to give more context on why the call was dropped or is reconnecting.
1.0.2 - 2025-02-14 #
Bug Fixing #
- Fixed an issue where the call states were not being updated correctly, identifying a call as active when it was still connecting (ICE Gathering). This caused a scenario where users thought a call was active but couldn't hear anything.
1.0.1 - 2025-02-10 #
Bug Fixing #
- Fixed an issue where running stats on calls could cause a call to end if the description was not set.
- Fixed the web version of the SDK that was having issues working on Safari.
1.0.0 - 2025-01-29 #
Enhancement - Breaking Changes #
- Call ID no longer required when ending call or using DTMF. As these methods belong to a call object, the call ID is inferred from the call object itself. This means users only need to keep track of the call objects that are in use and call the relevant methods on the call object itself.
Bug Fixing #
- Fixed an issue where the Bye Params (such as cause = USER_BUSY) were not being included in the ReceivedMessage.
0.1.4 - 2025-01-28 #
Enhancement #
- Update UUID dependency to latest version to avoid conflicts with implementers of the SDK
0.1.3 - 2025-01-06 #
0.1.2 - 2024-12-20 #
Bug Fixing #
- Fixed an issue where, when accepting a an invite, the destination number was being set to name instead of number.
0.1.1 - 2024-12-12 #
0.1.0 - 2024-11-07 #
Enhancement #
- Implemented WebSocket and RTC peer reconnection logic to ensure seamless recovery during network disconnects or switches.
0.0.18 - 2024-09-25 #
Bug Fixing #
- General bug fixes related to imports and how they work and switch between Mobile / Web
0.0.16 - 2024-08-01 #
0.0.15 - 2024-06-27 #
Bug Fixing #
- Fixed disposal of audio tracks to release iOS/Android microphones properly.
0.0.9 - 2022-10-22 #
0.0.8 - 2022-10-05 #
0.0.5 - 2022-09-02 #
0.0.4 - 2022-08-26 #
Bug Fixing #
- Improved stability by providing a session ID to the backend immediately rather than waiting to set one.
- Fixed gateway retry errors that continuously retried without a timeout.