Features of VoIP SIP SDK
Screenshots of VoIP SIP SDK Softphone
Quick facts about standard version of VoIP SIP SDK:
- All versions of Windows OS
- Support of SIP, RTP, and SDP
- STUN and other IETF standards (compatible with SER, OpenSER and Asterisk)
- Encryption ON THE BASIS OF SRTP, DLS, and DTLS (NEW)
- VoIP SDK ActiveX, COM, and plain dll files support
- SIP Registrar (SIP server) support
- Auto gain controller (AGC)
- Adaptive silence detection
- Adaptive jitter buffer
- DTMF (Dual Tone Multi Frequency)
- SIP Proxy authentication
- Support of calls: Hold/Retrieve Call, Forward Call (Blind Call Transfer), Transfer Call (Attended Transfer)
- Software volume control
- Record conversation into file
- Playback of WAV files into conversation
- IM interface
- Multi-party voice conference
- Multiline support
- Noise reduction
- Adaptive silence detection
- Mute Sound
- Support of Virtual Private Network (VPN)
- Noise reduction (supression)
- Media status support
- Getting of used codec per line
- Custom ringtone support (wav format)
- Redirection of calls to other phone lines
- Configuration saving and loading
- TCP support
- Selecting of media input and output devices
- Auto-answer
- Log file on/off setting
- NAT and firewall support
- Support of all kinds of Internet connections
- Prepared samples for programming languages (C#, C++, VB.NET, VB 6.0 and Delphi)
- IVR, call center, dialer ready
- SIP camera
- Call history and address book ready
- Royalty free licensing
- No yearly/monthly fee
Available codecs:
- G.711-ALaw
- G.711-muLaw
- G.726 (16k 24k 32k 40k)
- G.729A
- G.723
- iLBC
- LPC-10
- Speex (Narrow, Wide)
- L16
- RFC4733 DTMF tones
- GSM