v3.5.0

Enhancement

• WebRTC Call Stats and Troubleshooting Tool with JSON export and automatic upload to voice-sdk-proxy (WEBRTC-3227)
• Add ICE candidate pair details to Call Stats including connection state, network type, and candidate info (WEBRTC-3346)
• Add transport stats (dtlsState, iceState, srtpCipher, tlsVersion) to call report intervals (WEBRTC-3415)
• Add audioLevelAvg for inbound/outbound audio per stats interval (WEBRTC-3414)
• Add SDK latency measurement for WebRTC call establishment with milestone tracking (WEBRTC-3276)
• Handle telnyx_call_control_id in answer for Call Control integration (WEBRTC-3341)

Bug Fixing

• Fix clientState truncation when payload exceeds 57 bytes due to Base64.DEFAULT newline insertion (ENGDESK-50462)
• Prevent SIGSEGV crash in WebRTCReporter stats timer after PeerConnection teardown

v3.4.1

Enhancement

• Add conversation_id parameter to anonymous login methods for joining existing conversations (WEBRTC-3319)

Bug Fixing

• Fix CallState.CONNECTING emitted on active call when receiving a second incoming call

v3.4.0

Enhancement

• ICE Trickle Support for faster call setup
• Port TURN/STUN server configuration changes with custom ICE servers support
• Add DebugDataCollector for call debug logging
• Call Connection Benchmarking support
• Add answered_device_token parameter for push notification call answering
• Add DEV_TURN and DEV_STUN constants for development environment
• Support for starting calls with muted microphone (Invite/Answer muted)

Bug Fixing

• Fix incorrect client ready state on failed login
• Retry with exponential backoff when server closes during reconnection

v3.3.0

Enhancement

• Support for base64 encoded images in AI assistant messages
• Added AudioConstraints data class on invite and answer to allow for echoCancellation, noiseSuppression and autoGainControl

Bug Fixing

• When TxSocket is now closing by close() instead of cancel() to prevent onError callback

v3.2.0

Enhancement

• Connection State Exposure with DISCONNECTED, CONNECTED, RECONNECTING, CLIENT_READY states
• Expose Socket Connection Quality
• ICE Candidates Renegotiation

Bug Fixing

• Preferred Audio Codec Implementation

v3.1.0

Enhancement

• Anonymous Login + AI Agent related features
• Preferred Codec implementation

v3.0.0

Enhancement

• Implemented forceRelayCandidate parameter in Android WebRTC SDK.
• Migration from LiveData to Flows. (LiveData still supported for backward compatibility however methods are marked as deprecated)

v2.0.2

Enhancement

• Add Region parameter available to be set. This allows developers to restrict connection only to some regions.
• Add parameter for prefetching ice candidates during call initiation.

Bug Fixing

• Fixed an issue where the Termination Cause was always 'USER_BUSY' regardless of current call state. Now, when terminating an active call, the state will be 'NORMAL_CLEARING' and when rejecting an invite, the Termination Cause will be 'USER_BUSY'.

v2.0.1

Enhancement

• Further enhance error handling by modifying SocketObserver's onError method signature to include the Error Code as well as the Message.

v2.0.0

Enhancement

• Expose Call Termination Reasons in SDK and Surface Error Messages and Codes outside of the SDK. This allows developers to handle call termination reasons and error messages more effectively in their applications.