A powerful and highly versatile VoIP SDK to accelerate development of any type of VoIP-enabled application, like e.g. a SIP soft phone, teaching tool, live support, meeting tool or any other type of application which requires users being able to talk to each other.

Deliver SIP-based communications and services for PC-to-Phone, Phone- to-PC and PC-to-PC services and is fully inter-operable with any RFC SIP 3261

Provide the documentation and samples you need to integrate with other applications or websites, can be used by any development environment has ActiveX support.

Standard Features

• Multi phone lines.
• Caller ID.
• Call rejection.
• Call waiting.
• Call forward & blind transfer.
• Call hold.
• Call recording.
• Microphone and Speaker volume control with mute support.
• Microphone and Speaker visualization support.
• Redial, Auto answer & Do not disturb.
• Direct IP to IP Calling.
• Video call.

Enhanced Features

• Separate Call history for each registered user.
• Audio & Video tuning wizard.
• Acoustic echo cancellation, redundant audio coding, dynamic jitter buffer and adjustment, automatic gain control, voice activity detection.
• Support for the following audio codecs:

G.711, G.722.1, G.723, GSM, DVI4 and SIREN

• Support for the following video codecs:

            H.261 and H.263.

• Automatic selection of the best codec based on the other party’s capability, available bandwidth, and network conditions, switches the codec within a call in response to changing network conditions.

• UPnP-enabled NAT traversal.
• DTMF support.
• Live update.
• Fully commented sample applications for VC++, C#, VB.Net and JavaScript (Web demo).