Release Notes for Mobile SDK 5.2.48¶

Issues Fixed¶

General¶

• #2031 - prevent PJSIP crash due to double destroy of transport
• Add RTCP feedback for Opus encoder - this should improve audio quality under low bandwidth conditions.
• Add support for WebRTC "intelligibility enhancer" audio processing module configuration (that can now be triggered by the configuration server)
• Fixed an issue where the audio route was being set and reset multiple times during a call startup due to the audio device configuration being propagated too late into the call state.
• Debug logs now show the length of the SIP password used in the account configuration, to help debug issues with the account settings not being configured properly.
• There is no longer a warning log when receiving RTP after receiving SIP 200 OK and before sending the ACK - this should happen a lot for Registration-Free pickup calls and is not a problem.

Android¶

• Update dial*() APIs to match iOS implementation of returning a boolean value to let the caller know if the dial command was accepted and they should expect an onCallState(CALL_STATE_STARTING) event.
• Update AudioRoute enum to include the HEADPHONES value to match the iOS and core implementations. It is not used currently but it should still be available for consistency.
• Support a configuration server setting of a non-earpiece audio route as the default audio route, other than "earpiece", for use in non-phone devices.
• Fixed an occasional failure to properly re-init the SDK after shutdown.
• Fixed reading boolean account settings from cache.

Known Issues¶

• #426 - On Android, SDK cannot use NAT64 networks. This is a limitation of the operating system. Use dual-stack SIP servers to support IPv6 only clients.
• #135 - DNS SRV isn't supported.