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.
- 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.
- 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.
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