Home Documentation Features Samples Download Price Demo Contact Us   Frequently Asked Questions How to get a trial? How to develop a softphone? Does it work with Asterisk? What codecs does it support? Echo & noise cancellation supported? Does it support multi-user conference? Does it support call transfer & hold? I want to record the call conversation? Does it work on Windows 7 & Vista? more You can add many new and powerful softphone related features and just discover, how much better your application and webpages can be with:   Supported features: SIP Proxy authentication. ? Dial/Receive phone calls. ? Multi-lines support. ? Multi-party voice conference.? Line Hold.? Call transfer. ? Encrypt SIP account settings. ? Acoustic echo cancellation or suppression. ? Noise cancellation or suppression. ? AGC (auto gain controller). ? Record conversation into wave (.wav) file. ? Play wave (.wav) file to the remote end. ? Friendly to NAT and other firewalls. ? Keep-alive packets to NAT/firewall. ? Narrowband, wideband voice codecs. ? Adaptive Jitter buffer. ? Packet Loss Concealment. ? DTMF tones generation. ? DTMF tones detection. ? Do Not Disturb (DND). ? Mic & Speaker Volume. ? Microsoft Autenticode Certificate. ? Works with all kind of Internet connections. ? Free product version upgrades. ? SIP PROXY AUTHENTICATION     [ Back to Top ] VaxVoIP SIP SDK enables to register with the SIP proxy server by providing Login Id and Login password.   DIAL/RECEIVE PHONE CALLS     [ Back to Top ] You can dial and receive phone calls through any SIP based server, gateway or ITSP (Internet Telephony Service Provider).   MULTI-LINES SUPPORT     [ Back to Top ] VaxVoIP SIP SDK enables to initialize the component with a user-define specific number of lines. You will be free to start the component with 4, 8, 10, 20, 40, 80 or more number of lines. Such feature is use to start conference call, consult call transfer, dial/receive multiple phone calls and for many other purposes.   MULTI-PARTY VOICE CONFERENCE     [ Back to Top ] User can dial and receive multiple calls to start conference call.   LINE HOLD     [ Back to Top ] During the call session, user can put any line on hold.   ENCRYPT SIP ACCOUNT SETTINGS     [ Back to Top ] If you hard-code the SIP account settings in your webpage then any user can easily view those settings by viewing the source of that webpage. To prevent such situation, VaxVoIP allows you to encrypt the hard-coded/static SIP account settings and then use them in your webpages. For more details, please see Encrypt SIP account settings in the documentation section.   CALL TRANSFER     [ Back to Top ] Transfer a call to other phone number, sip account or sip uri.   ACOUSTIC ECHO CANCELLATION OR SUPPRESSION     [ Back to Top ] In order to eliminate the acoustic feedback an echo canceller is introduced in the VaxVoIP SIP SDK. Hands-free or Internet telephony imposes several problems. The principal one is due to the coupling between loudspeaker and microphone. The loudspeaker signal is echoed back to the microphone and transmitted back to its origin. As a result the far-end participant perceives this as an echo. Note: Acoustic Echo Cancellation (AEC) works under cetain conditions. If your hardware or environment does not fullfil any of those conditions then there is chance that AEC (Acoustic Echo Cancellation) may not work properly.   NOISE CANCELLATION OR SUPPRESSION     [ Back to Top ] VaxVoIP SIP SDK offers advanced Noise Cancellation technology that allows significant suppression of any background noise and produce high quality of output speech.   AGC (AUTO GAIN CONTROLLER)     [ Back to Top ] We support AGC (auto gain controller). AGC is a mechanism by which input voice gain/volume is adjusted automatically based on input signal level.   RECORD CONVERSATION INTO WAVE (.WAV) FILE     [ Back to Top ] During the phone call, you will be able to record the conversation into wave (.wav) file for later play back.   PLAY WAVE (.WAV) FILE TO THE REMOTE END     [ Back to Top ] VaxVoIP SIP SDK export methods to play wave (.wav) file to the remote end.   FRIENDLY TO NAT AND OTHER FIREWALLS     [ Back to Top ] User can set SIP outbound proxy inorder to make and receive phone calls behind the NAT/firewall. In some cases, ITSP (Internet Telephony service provider) support outbound proxy. Outbound proxy is a way to let the NAT/firewall user make and receive phone calls. If the NAT/firewall router does not support SIP pass-through, you need to consult your ITSP if they support SIP outbound proxy. Since different NAT router vendor implement NAT differently. Typically ITSP may provide SIP outbound proxy to resolve NAT pass-through issues. STUN is not a good idea to support NAT pass-through, because STUN does NOT support symmetric NAT type, symmetric NAT is more secure and widely use for commercial purposes. Almost all branded routers support symmetric NAT type, even Microsoft windows SERVER 2000 & 2003 built-in NAT is also base upon symmetric NAT type. Please see STUN RFC for more details.   KEEP-ALIVE PACKETS TO NAT/FIREWALL     [ Back to Top ] VaxVoIP SIP SDK support keep alive feature. When you enable it, VaxSIP component starts sending keep-alive packets and keeps the port open at firewall ends.   NARROWBAND & WIDEBAND VOICE CODECS     [ Back to Top ] VaxVoIP SIP SDK support for both narrowband and wideband codecs that's why it works with all type of Internet connections. G.729 G711 U-Law G711 A-Law iLBC GSM 6.10   ADAPTIVE JITTER BUFFER     [ Back to Top ] Jitter buffers are used to smooth delay variations in received audio by buffering the packets and adjusting their rendering. The result is a smoother delivery of audio to the user.   PACKET LOSS CONCEALMENT     [ Back to Top ] Packet Loss Concealment (PLC) is a technique used to mask the effects of lost or discarded packets. PLC is generally effective only for small numbers of consecutive lost packets, for example a total of 20-30 milliseconds of speech, and for low packet loss rates.   DTMF TONES GENERATION     [ Back to Top ] VaxVoIP SIP allows applications and webpages to generate DTMF tones.   DTMF TONES DETECTION     [ Back to Top ] VaxVoIP SIP SDK also support DTMF tones detection feature.   DO NOT DISTURB (DND)     [ Back to Top ] VaxVoIP SIP SDK support DND (Do Not Disturb) feature.   MIC & SPEAKERS VOLUME     [ Back to Top ] User can control Mic and Speakers volume direclty.   MICROSOFT AUTHENTICODE CERTIFICATE     [ Back to Top ] The VaxVoIP component is already signed by Microsoft Authenticode Certificate. So you don't need to purchase any Code Sign Certificate before using VaxVoIP activeX or VaxVoIP client-side downloadable controls.   WORKS WITH ALL KINDS OF INTERNET CONNECTIONS     [ Back to Top ] Due to the support of both NARROWBAND & WIDEBAND voice codecs, VaxVoIP component works with all kinds of Internet connections Dialup modems, ADSL, Cable Modems etc.   FREE PRODUCT VERSION UPGRADES     [ Back to Top ] After purchasing the license key, you will get the product new versions and upgrades free of charge.   Some of our customers   Supported Languages Visual Basic .NET Visual C++ .NET Visual C# .NET Visual Basic Visual C++ Borland Delphi Borland C++ JavaScript/HTML    Become a Reseller VaxVoIP offers resellership program to those companies, who are looking for business opportunity and want to become a reseller for VaxVoIP SIP SDK. If you want to become a reseller, please contact contactus@vaxvoip.com    Custom  Development VaxVoIP offers affordable software customization for our products. Whether you need simple changes or need new functionality, we offer services to meet your needs. If you are interested in custom software development, please contact customdev@vaxvoip.com.   Terms Of Use Privacy Policy   Copyright © 2010 VaxSoft. All Rights Reserved.