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Our VoIP SIP SDK provides a powerful and highly customizable solution to quickly add SIP based dial and receive phone calls features in your software applications. It accelerates the development of SIP compliant soft phone with a fully-customizable user interface and brand name. The Abto LLC VoIP 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 intranet networks delivering superior voice quality. It supports DTMF, adaptive silence detection, adaptive jitter buffer! Abto LLC VoIP SIP SDK is based on IETF standards (SIP, STUN, etc.), so it should be compatible with other standard based products such as Asterisk, GNU Gatekeeper other.
The new Version 3.0.11 comes now with:
- Added method IVOIPManager::put_SamplesPerFrame /get_SamplesPerFrame
- Added method IVOIPManager::put_DialToneEnabled /get_DialToneEnabled
- Extended DTMF tones support
- New method SetConnectionContribution(int partHandle, int inputGain, int outputGain)
- Added new method IAbtoPhone::IsLineOccupied
- Modified 'SetSipProxy' method, now it allows to set empty proxy
- Corrected code that works with networks interfaces (IPv4/v6 suffix added, transport selection code modified)
- Added new property put_AdditionalDnsServer /get_AdditionalDnsServer (on both interfaces) - Allows to add AdditionalDnsServer like google 8.8.8.8
- Added new method 'TransferConnection(connId, addr)' - Allows transfer selected connection from comf call
- Help Updated
The new Version 3.0.6 comes now with:
- Manual network interface selection
- New examples:
1. "MlSampleGSMGateway" that allows route incoming calls to GSM phone though bluetooth
2. "MlSampleOperator"
Can be used as call center operator terminal.
Made as auto-answer phone
Allows for manager connect to operator and listen conversation
The new Version 3.0.5 comes now with:
- AuthID handling
- New config variable "LocalIP"
The new Version 3.0.4 comes now with:
- Explicit multilines handling added
- Conferences quality issues fixed
- Configuation handling enhanced
- Stability has been greatly improved
- New samples added
- Help updated
- Samples updated
The new Version 2.1.4 comes now with:
- Added FrameReceived event
- Added saving codec checked state when changing codec order
- Added .Net wrapper
- Added C++ Windowless Sample
- Removed Set/GetListenPort functions. Using property instead
- Help updated
- Samples updated
The new Version 2.1.3 comes now with:
- Added DLL only version, along with ActiveX
- C# sample for DLL version
- C# Console version updated
- C++ application update
- SDP updates
- Conference calls updated
- Caller-id and user-agent customization
- Retrieving external IP addres from STUN server
The new Version 2.1 comes now with:
- IM interface added
- fixed issue with RFC4566
- Codec disabling
- Help updated
- Samples updated
- Playback of different formats of WAV files
- Software volume control
The new Version 2.0 comes now with:
- Noise Reduction option
- Auto Gain Control option
- Added local number parameter
- Jitter buffer parameters
- No forms event rising problem fixed
- Samples updated
- Help updated
The new Version 1.6 comes now with:
- FAX recognition
- Conference support introduced
- Added new VC6 sample
- Samples updated
The new Version 1.5 comes now with:
- fixed STUN support
- added support of Authorization Id
The new Version 1.4 comes now with:
- Mute Sound
- VPN support
- Fixed VB6 compilation issue
The new Version 1.3 comes now with:
- Call transfer support
- Added status code on call reject
- Added Unauthorized and Trying events for user registration
- Fixed bug with default sound device
- Fixed to be recognized as valid ActiveX component
- Samples updated
The new Version 1.2 comes now with:
- Added Hold/Retrieve
- Added Record conversation into WAV
- Delphi sample updated
- VB sample updated
- HTML Javascript sample updated
- Extended OnIncomingCall event arguments
The new Version 1.1 comes now with:
- Added abitility to play wav files into conversation
- New event OnPlayFinished to know when play is finished
System Requirements: WinXP/2k/Vista/7.
Key words: VoIP SDK, SIP SDK, soft phone.
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