VoIP SIP SDK

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 intern networks either by speaking and delivers 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, OpenSER other.

Dynamically loadable codecs

Registrar support

STUN support

Samples on Delphi, C#, VB, VB.NET, C#, HTML

echo cancelation

support of playing audio files

Hold/Retrieve with sample

Record conversation into WAV

Delphi sample updated

VB sample updated

HTML JavaScript sample updated

Extended OnIncomingCall event arguments

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

Mute Sound

VPN support

Fixed VB6 compilation issue

Support of Authorization Id

FAX recognition

Noise reduction

Auto gain

Local number parameter

Help updated

Jitter buffer parameters

Windowless samples on C++ and .NET

The new Version 3.0.4 comes with:

* Explicit multilines handling added

* Conferences quality issues fixed

* Configuation handling enhanced

* Stability has been greatly improved

* New samples added

* Help updated

* Samples updated