ABTO SIP SDK 3.0.12
BTO SIP SDK provides a solution to quickly build VoIP softphone that can dial and receive calls on your computer or add VoIP features into your software or web website.
Our powerful and highly customizable VoIP client designed as ActiveX control and has wide range of Features: fully-customizable user interface and brand name, conference call, multiline features, ability to send text messages, dynamically loadable codec's, DTMF support, adaptive silence detection, adaptive jitter buffer, work through firewall or NAT, and more.
Also ABTO SIP SDK is based on IETF standards (SIP, STUN, etc.) so it is compatible with standard registrars such as Asterisk, Elastix, 3CX etc.
The new Version 3.0.11 comes with:
* Support of all versions of Windows OS
* Support of SIP, RTP, and SDP
* STUN and other IETF standards (compatible with SER, OpenSER and Asterisk)
* Encryption mode that accepts secure media
* Support of ActiveX, COM, and plain dll files
* SIP Registrar (SIP server) support
* Auto gain controller (AGC)
* Adaptive silence detection
* Adaptive jitter buffer
* DTMF (Dual Tone Multi Frequency)
* SIP Proxy authentication
* Support of calls: Hold/Retrieve Call, Forward Call (Blind Call Transfer), Transfer Call (Attended Transfer)
* Software volume control
* Record conversation into file
* Playback of WAV files into conversation
* IM interface
* Multi-party voice conference
* Multiline support
* Noise reduction
* Adaptive silence detection
* Mute Sound
* Support of Virtual Private Network (VPN)
Available codecs:
* G.711-ALaw
* G.711-muLaw
* G.726 (16k 24k 32k 40k)
* G.729A
* iLBC
* LPC-10
* Speex (Narrow, Wide)
* L16
* RFC4733 DTMF tones
* GSM
Requirements
Changes: 3.0.12
Manual selection of network interface for RTP data exchange.
