ABTO VoIP SIP SDK 4.5

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ABTO 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.

The ABTO SIP SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software).

It enables a worldwide communication over the internet or intern networks either by speaking and/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller, acoustic echo suppression, noise suppression, reverb suppression and mute microphone/speaker for each line.
 
It supports multiple lines, call hold, call transfer, DTMF, adaptive silence detection, adaptive jitter buffer, record and playing WAV!

The ABTO SIP SDK provides the documentation, samples and related libraries you need to integrate with other applications or websites.

The ABTO SIP SDK is designed as ActiveX simple object (without GUI) to be used by Automation clients from any programming language that support this feature.

SDK includes all of the necessary software components, source code examples, explanations as well as necessary service programs, libraries and components.

With the help of ABTO VoIP SIP SDK the task of developing any type of VoIP-related application is much faster and easier.

NOTE: Users will need to request a trial license in order to use the software. Here are some key features of "ABTO VoIP SIP SDK":

· Support of SIP, RTP, and SDP
· STUN and other IETF standards (compatible with SER, OpenSER and Asterisk)
· Encryption ON THE BASIS OF SRTP, DLS, and DTLS (NEW)
· VoIP SDK ActiveX, COM, and plain dll files support
· 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)
· Noise reduction (supression)
· Media status support
· Getting of used codec per line
· Custom ringtone support (wav format)
· Redirection of calls to other phone lines
· Configuration saving and loading
· TCP support
· Selecting of media input and output devices
· Auto-answer
· Log file on/off setting
· NAT and firewall support
· Support of all kinds of Internet connections
· Prepared samples for programming languages (C#, C++, VB.NET, VB 6.0 and Delphi)
· IVR, call center, dialer ready
· SIP camera
· Call history and address book ready
Requirements:

· Sound Card
· Microphone
Limitations:

· 10-day trial What's New in This Release: [ read full changelog ]

· Fixed playing dial tones
· Added new method GetSIPHeaderValueLine
· Added ability to use method GetSIPHeaderValue on outgoing messages too
· Added ability to handle NOTIFY messages event when SDK is not subscribed for them.
· Added ability to record microphone sound only. StartRecordingConnection(0);
· Fixed wrong G729 codec file in installation package
· Added ability to use fixed transport in Via header
· Added ability to automatically detect sound output device channel count
· Fixed bug with ‘From’ header in outgoing text messages
· Changed SDK version in dll resources
· Added ability to change ringtone depending on the incoming call
· Added advanced log on selection camera format and selection source interface
· Fixed selection source interface in WinCompat::determineSourceInterface
· Added new feature ‘muLtiple sip accounts registration’
· Added ability to enable\disable video call feature without restarting phone