Update VoIP SIP SDK for Windows

Download an important update of VoIP SIP SDK for Windows:


Change log:

– Improved handling errors in hold processing and raising event ‘OnHoldError’;

– Fixed cleanup call state – when call put on hold and ended SDK restores line state;

– Added ability to extract the full name of audio device without 32 characters limitation;

– Fixed bug with the port number when STUN enabled and app makes multiple outgoing calls;

– Added new method ‘SetTonesContribution’;

– Improved sending Video RTP. Now first 5 frames are forced as ‘key frames’,
which allows remote side to display received video almost immediately;

– Fixed bug in Unhold video implementation;

– Modified code, which configures TLS server mode (ability to accept incoming requests);

– Fixed handling 183 without ‘To.tag’;

– Redesigned existing AEC implementation based on Speex;

– Fixed bug with ending call after un-successful transfer;

– Added new method ‘StartCallAcc’ allowing to make outgoing call from selected account;

– Added new method ‘SetPlayingFileContribution(LONG gain, int connectionId)’ allowing setting file player gain per connection;

– Added new AEC mode based on Webrtcm.
To enable it set on initialization stage:
phoneCfg.EchoCancelationEnabled = 2;
phoneCfg.SamplesPerSecond = 16000;

– Set default keep alive timer 60 seconds;

– Added new method ‘GetVideoFrameSize’.
Returns size of local/remote video frame during video call.