http://voipcore.com
SIP stack
SIP Stack(completely own and developed by PCBest. We can decode any additional features in SIP message for your SIP Project, or fix any problems that may exist in the SIP core)
Very stable and compact size
Compatible SIP Servers, Proxy and PBX Full compatible with Open SER, Asterisk, Cisco CallManager, Audio Codes, 3CX, Radvision, Rainbow and more others SIP platforms.
Compatible SIP Hardwares Full compatible with DLink, Audio codes, Grandstream, Cisco, Huawei, other major SIP hardware phones and PBXs.
Programming interfaces
C++ head files and lib
.NET assembly(managed interfaces)
ActiveX control
Standard DLL Interface
Supported development tools
MS Visual Studio 2003/2005/2008(C#, VB.NET, J#, ASP.NET,...)
MS Visual Studio 6(VC6, VB6, ...)
Borland C++ 5/6/7 Delphi 6/7 CodeGear Delphi 2007 CodeGear C++ Builder 2007
Java, JavaScript, HTML, and other windows development tools which support ActiveX control
Audio call
Yes
Audio codecs
G.711-ALaw,G.711-muLaw,G.726 (16k 24k 32k 40k),G.729A,iLBC,LPC-10,Speex (Narrow, Wide),L16,RFC4733 DTMF tones,GSM.
Basic Telephony
Hold, Transfer, Do Not Disturb(DND), Auto answer, Redial, Redirect Call(302)
Conference
YES
Very powerful server feature
Voice Activity Detection(Human or Answer machine detection)
YES
Record(Dynamically turn on during a live call)
YES (Record Audio Mix )
Record the audio data and save as WAV files
Wav file play and record YES(Support .wav and .au files)
Audio format can be:
8K 16bit mono PCM
8k 8bit mono mulaw/alaw Play a WAV file instead of microphone, or record incoming voice into a WAV
Music On Hold YES
Message Waiting Indicator (MWI) YES Implemented as RFC 3842
Supported SIP Methods
REGISTER, INVITE, CANCEL, INFO, BYE, ACK, REFER, SUBSCRIBE, OPTIONS, NOTIFY, MESSAGE, UPDATE
RFC supported
RFC 3261, RFC 3665, RFC 2833, RFC 2327, RFC 3264, RFC 3550, RFC 3263, RFC 3891,RFC 3515, RFC 3420, RFC 3892, RFC 3265, RFC 3666, RFC 3489, RFC 3920, RFC 3921, RFC 3922, RFC 3923, RFC 4622, RFC 4854, RFC 4979, RFC 3842,
Authentication
HTTP Basic
Digest Authentication
DTMF supported
DTMF Detection and Sending
RFC2833 / SIP INFO / Inband / Auto
***** All possible DTMF methods. Set auto to use RTP(RFC 2833) or inband audio depends on SDP exchanges
Multiple Concurrent Calls
Yes
Support dynamically change sound devices during a live call
YES
*****Very powerful feature. Good for call center agent softphone to switch between Speaker and USB headset without cutting off a call
RTP Package Access
Support access incoming and outgoing RTP audio stream directly. And support change RTP audio stream to integrate TTS and ASR engine
Very powerful server feature
DirectX Audio Stream Access
Yes. Can Access and change the DirectX audio on the middle way on both play and record direction
Very powerful server feature
Tone Detection
Yes.
Very powerful server feature
Microphone & Speaker Device Selector
YES
Microphone & Speaker Volume control
YES
SIP UDP Support
YES
Acoustic Echo Cancellation
YES
Adaptive Jitter buffer
YES
PLC (Packet Lost Concealment)
YES
AES (Acoustic echo cancellation or suppression)
YES
Noise cancellation or suppression
YES
Outbound Proxy supported
YES
STUN supported
YES
Jitter Buffer
YES
Free product version upgrades
YES
Private Encrypt
YES
Call History
YES
Address book
YES
Channel Timer
YES
GUI customization
YES
http://voipcore.com