VoIP SIP Client SDK full changelog
VoIP SIP Client SDK 1.3
released Jul 8, 2010 (Major Update)
* g729 and g723 Codecs support
* Multiple and single Codec selection support
* Failure codes support (get SIP Message Response Code, SIP Message Response Text)
* RTP/RTCP Port setting (for inbound RTP traffic)
* Reduce audio latency and audio latency settings
* Multiple and single Codec selection support
* Failure codes support (get SIP Message Response Code, SIP Message Response Text)
* RTP/RTCP Port setting (for inbound RTP traffic)
* Reduce audio latency and audio latency settings
VoIP SIP Client SDK 1.2
released Nov 17, 2009 (Major Update)
*Easily make and receive SIP based phone calls
* Multi-Line, Multi-User conference support
* Make and receive SIP based phone calls
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A/U, Speex, GSM6.10, iLBC, L16, g729)
* STUN, DTMF and Call Transfer
* Multi-Line, Multi-User conference support
* Make and receive SIP based phone calls
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A/U, Speex, GSM6.10, iLBC, L16, g729)
* STUN, DTMF and Call Transfer
VoIP SIP Client SDK 1.1
released Oct 21, 2009 (New Release)
*Easily make and receive SIP based phone calls
* Multi-Line, Multi-User conference support
* Make and receive SIP based phone calls
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A/U, Speex, GSM6.10, iLBC, L16, g729)
* STUN, DTMF and Call Transfer
* Multi-Line, Multi-User conference support
* Make and receive SIP based phone calls
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A/U, Speex, GSM6.10, iLBC, L16, g729)
* STUN, DTMF and Call Transfer
VoIP SIP Client SDK 1.0
released Sep 3, 2009 (New Release)
*Easily make and receive SIP based phone calls
* Multi-Line, Multi-User conference support
* Make and receive SIP based phone calls
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A/U, Speex, GSM6.10, iLBC, L16, g729)
* STUN, DTMF and Call Transfer
* Multi-Line, Multi-User conference support
* Make and receive SIP based phone calls
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A/U, Speex, GSM6.10, iLBC, L16, g729)
* STUN, DTMF and Call Transfer
VoIP SIP Client SDK for Windows 7 - free download notice
Windows 7 Download periodically updates software information of VoIP SIP Client SDK from the publisher.
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