Update VoIP SIP SDK for Windows

Please download our latest VoIP SIP SDK for Windows.

Change log:

– Added ability to dial remote side with empty user name (dial IP address only)

– Fixed potential crash when app invokes ‘RestartNetworkSubsystem’ and SIP message received

– Fixed potential crash when app invokes ‘ApplyConfig’ and other methods from different threads.

– Improved log output

– Added ability to send video from newly connected camera

– Added code, which prevents using incorrect default IP when SDK starts without registration

– Added code which verifies stack restart condition, based on ‘auto’ addr

– Added code which properly handles Unicode device names

– Fixed bug with Proxy, added code which restarts stack when proxy changed

– Fixed bug unregister on exit

– Added new property ‘OverridePublicIP’

– Added new property ‘rportEnabled’

– Fixed bug when app tries to invoke ApplyConfig few times, before initialization

– Added code which generates name for crash.dmp. Now it contains creation time and SDK version.

– Added new event ‘OnReceivedSipNotifyMsgInDialog’

– Added ability to set own headers for REGISTER request

– Improved handling in-dialog NOTIFY – answer with 200 OK

– Fixed playing wav files encoded with G711ALaw (compressionType=6)

– Added ability to set different Mic/Spkr volume for each phone instance

– Added new setting ‘AllowedRemoteIP’ (When set ip or ip:port SDK handles only packets from this source)

– Fixed possible crash with RTCP when user put call on hold

– Added ability to automatically stop dial tone, when received RTP from remote side

– Fixed potential crash when app switches account during the call

– Added new setting ‘TlsForceSipScheme’

– Fixed bug with STUN caused by adding new setting ‘OverridePublicIP’

– Fixed sending IM (now SDK uses proper account)

– Added code which prevents from corrupting stored SIP message string

– Fixed possible crash when app receives incoming call after adding new accounts

– Added more log output for start/stop audio devices

– Fixed ‘sip’ scheme in ‘Contact’ header when ‘sips’ enabled

– Fixed possible crash with IM sending

– Fixed H264 decoder initialization when received empty/wrong ‘fmtp’

– Redesigned code which loads codec libraries (Unicode, longer then 255chars)

– Redesigned file recorder (‘lame_enc.dll’ is not required anymore)

– Added ability to use ListenPort as outgoing TCP port

– Fixed potential crash with handling DTMF events