SIP SDK, VoIP SDK | PCBest Networks


You don't have to experience on so many Telecom and Internet protocols and concepts. The SIP SDK has integrated everything for you. STOP Struggling in SIP messages, call status, and multi-threadings. This pure OOP designed SIP SDK will let you focus on your business logic.

This VoIP SDK is a pure Host Media Process(HMP) engine. It allows you to develop VoIP application without any hardware(boards), if you are a IVR or CTI developer, and trying to migrate your board application to boardless.

You can also use PCBest SIP SDK to develop your own customized SIP softphone for PBX. Lots of softphone samples provided in SDK package.

Why choose PCBest SIP SDK?

Download PCBest Networks SIP SDK v2.10e x86(Feb 04, 2018 release) - Added TCP support for SIP protocol. Added proxy samples for C# and VB.NET. Updated C# and VB.NET softphone sample, tone detection, conference audio and whisper, add your own SIP header, and other powerful features. with proxy BLF support. with DNS SRV support. Enhanced performance. More CPS supported.

Download PCBest Networks SIP SDK v2.10e x64(Feb 04, 2018 release)

This SDK can be used to develop SIP softphone or SIP server applications, but it has two different licenses for your applications. Check Price List or Buy SIP SDK

Please be noted: The application developed on trial version of SDK can only run for one hour each time, and each call is limited to 3 minutes.

PCBest SIP SDK API Reference(pdf)
Online Guide for SDK Configuration Items


SIP SDK Specifications:


SIP stack

PCBest Networks 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.  

Supported Platforms

MS Windows(2000/XP/2003/Vista/2008)                           


Programming interfaces

C++ head files and lib
.NET assembly(managed interfaces)
ActiveX control
Standard DLL Interface

Supported development tools

MS Visual Studio 2003/2005/2008/2010/2012(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


Audio codecs

G.711 uLaw/aLaw, G726, GSM, iLBC, Speex. G729(optional).


Basic Telephony

Hold, Transfer, Do Not Disturb(DND), Auto answer, Redial, Redirect Call(302)




Very powerful server feature

Voice Activity Detection(Human or Answer machine detection)



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



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, 



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



Support dynamically change sound devices during a live call


*****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


Very powerful server feature

Microphone & Speaker Device Selector



Microphone & Speaker Volume control



SIP UDP Support



Acoustic Echo Cancellation



Outbound Proxy supported



STUN supported



Jitter Buffer



Free product version upgrades


We provide 3 months free upgrade

Private Encrypt



Call History



Address book



Channel Timer



GUI customization