Dialup	(56k)	-	20 m 37 s
ISDN	(128k)	-	09 m 02 s
DSL	(512k)	-	02 m 16 s
Cable	(1024k)	-	01 m 08 s
T1	(1484k)	-	47 s

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.

v1.3
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

v1.4
Mute Sound
VPN support
Fixed VB6 compilation issue

Version 1.5
- fixed STUN support
- added support of Authorization Id

Version 1.6 :
- FAX recognition
- Conference support introduced
- Added new VC6 sample
- Samples updated

Version 2.0:
- Noise reduction
- Auto gain
- Added local number parameter
- Help updated
- Jitter buffer parameters
- Samples updated
- Windowless samples on C++ and .NET

Version 2.1:
- IM interface added
- fixed issue with RFC4566
- Codec disabling
- Help updated
- Samples updated
- Playback of different formats of WAV files
- Software volume control