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SIP stack
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PCBest Network 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)
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Very stable and compact size
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Compatible SIP Servers, Proxy and PBX |
Full compatible with Open SER, Asterisk, Cisco CallManager, Audio Codes, 3CX, Radvision, Rainbow and more others SIP platforms. |
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Compatible SIP Hardwares |
Full compatible with DLink, Audio codes, Grandstream, Cisco, Huawei, other major SIP hardware phones and PBXs. |
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Supported Platforms
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MS Windows(2000/XP/2003/Vista/2008)
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Programming interfaces
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C++ head files and lib
.NET assembly(managed interfaces)
ActiveX control
Standard DLL Interface
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Supported development tools
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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
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Audio call
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Yes
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Audio codecs
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G.711 uLaw/aLaw, G726, GSM, iLBC, Speex. G729(optional).
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Basic Telephony
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Hold, Transfer, Do Not Disturb(DND), Auto answer, Redial, Redirect Call(302)
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Conference
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YES
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Very powerful server feature |
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Voice Activity Detection(Human or Answer machine detection)
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YES
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Record(Dynamically turn on during a live call)
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YES (Record Audio Mix )
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Record the audio data and save as WAV files
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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 |
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Music On Hold |
YES |
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Message Waiting Indicator (MWI) |
YES |
Implemented as RFC 3842 |
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Supported SIP Methods
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REGISTER, INVITE, CANCEL, INFO, BYE, ACK, REFER, SUBSCRIBE, OPTIONS, NOTIFY, MESSAGE, UPDATE
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RFC supported
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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,
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Authentication
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HTTP Basic
Digest Authentication
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DTMF supported
DTMF Detection and Sending
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RFC2833 / SIP INFO / Inband / Auto
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***** All possible DTMF methods. Set auto to use RTP(RFC 2833) or inband audio
depends on SDP exchanges |
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Multiple Concurrent Calls
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Yes
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Support dynamically change sound devices during a live call
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YES
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*****Very powerful feature. Good for call center agent softphone to switch between
Speaker and USB headset without cutting off a call |
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RTP Package Access
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Support access incoming and outgoing RTP audio stream directly.
And support change RTP audio stream to integrate TTS and ASR engine
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Very powerful server feature |
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DirectX Audio Stream Access
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Yes. Can Access and change the DirectX audio on the middle way on both play and record direction
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Very powerful server feature |
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Tone Detection
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Yes.
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Very powerful server feature |
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Microphone & Speaker Device Selector
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YES
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Microphone & Speaker Volume control
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YES
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SIP UDP Support
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YES
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Acoustic Echo Cancellation
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YES
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Outbound Proxy supported
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YES
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STUN supported
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YES
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Jitter Buffer
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YES
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Free product version upgrades
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YES
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We provide 3 months free upgrade
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Private Encrypt
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YES
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Call History
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YES
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Address book
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YES
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Channel Timer
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YES
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GUI customization
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YES
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