OVERVIEWVaxVoIP WebPhone SDK provides tools and components to quickly add SIP (Session Initiation Protocol) based dial and receive phone calls, audio and video conferencing feature in your webpages. It accelerates the development of webphone having your own GUI (graphical user interface) and brand name.
VaxVoIP WebPhone SDK includes Server WebRTC library and html/jscript based webphone. It is really easy to incorporate VaxVoIP WebPhone SDK in your webpages. WebPhone sample source codes are available to download.
It delivers superior voice quality by integrating advanced digital voice processing features including acoustic echo cancellation, noise cancellation and adaptive jitter buffering. For more details, please visit the features link.
FEATURESAdd many new and powerful WebPhone related features and just discover, how much better your applications can be with:
VaxVoIP WebPhone SDK (Software Development Kit). It contains sample codes, technical manual, COM component and demo application. It is a complete package to develop a WebRTC based WebPhone.
VaxVoIP WebPhone SDK contains single COM (Component Object Model) component, and can be used with any application development tool, VC++, C#, VB.NET, Delphi, C++, etc.
Webphone developed by VaxVoIP SIP SDK works with the latest versions of FireFox and Google Chrome web browsers. Demo webphone is available for testing purposes.
VaxVoIP WebPhone SDK export funtions to activate SIP based video conference.
VaxVoIP WebPhone SDK provides functionality to develop call center coaching services. In which supervisor instructs to the agent in real-time. But customer does not hear the voice of the supervisor.
VaxVoIP WebPhone SDK enables to register with the SIP proxy server by providing Login Id and Login password.
One can easily add SIP based Instant Messiging and Presence feature in its VaxVoIP integrated WebPhone. VaxVoIP WebPhone SDK supports SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions) protocol.
SIMPLE is the SIP protocol extention to send and receive SIP based chat messages and status (online, offline, away, on the phone etc). So, it really easy to add and develop SIP based chat feature, please see the demo for more details.
You can dial and receive phone calls through any SIP based server, gateway or ITSP (Internet Telephony Service Provider).
Forward an incoming call to other phone number, user name or sip account.
Transfer a call to other phone number, user name, sip account or sip uri.
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 (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.
VaxVoIP SIP allows applications to generate DTMF tones.
VaxVoIP WebPhone SDK support DND (Do Not Disturb) feature.
After purchasing the license key, you will get the one year product new versions and upgrades free of charges.
HTML/JSCRIPT & SERVER APP
Visual cSharp .NET
Visual Basic .NET
Visual cSharp .NET (WPF)
Visual Basic .NET (WPF)
As we are happy to provide a 30 day free trial, please download any sample code and try it out.
WEBPHONE - DEMO (HOSTED) Launch Now
WEBPHONE - DEMO (HTML/JSCRIPT & SERVER APP) Download Now
VaxVoIP WebPhone SDK is a complete package to develop a fully functional WebRTC Server and Web based softphone. As we provide 30 days free trial, please download VaxVoIP WebPhone SDK and see for yourself.
SERVER WEB-RTC COM (.DLL)
VaxVoIP WebPhone SDK includes ServerWebRTC COM (Component Object Model) based component, that can be used to develop WebRTC Server using different development tools and languages. Please download WebPhone SDK, run the sample code and follow the steps.
VaxVoIP WebPhone SDK also provides html/jscript based WebPhone. WebPhone connects to ServerWebRTC and registers the provided SIP account.
We offer single product/software based license to use VaxVoIP WebPhone SDK. You are only limited to use one license with one of your product/software.
You will get one year of FREE technical support, new versions and upgrades with purchase of a license. After first year of free support, given below support packages can be opted any time.
SINGLE SERVICE LIMITED CONCURRENT CALLS LICENSE
20 CONCURRENT CALLS Buy Now -$860 USD
40 CONCURRENT CALLS Buy Now -$1,420 USD
80 CONCURRENT CALLS Buy Now -$1,880 USD
160 CONCURRENT CALLS Buy Now -$3,420 USD
320 CONCURRENT CALLS Buy Now -$5,480 USD
SINGLE SERVICE UNLIMITED CONCURRENT CALLS LICENSE Buy Now -$8,640 USD
3 MONTHS PAYMENT PLAN ($9,630 USD) Buy Now -$3,210 USD
6 MONTHS PAYMENT PLAN ($10,236 USD) Buy Now -$1,706 USD
12 MONTHS PAYMENT PLAN ($12,840 USD) Buy Now -$1,070 USD
UNLIMITED SERVICES UNLIMITED CONCURRENT CALLS LICENSE Buy Now -$12,620 USD
3 MONTHS PAYMENT PLAN ($14,280 USD) Buy Now -$4,760 USD
6 MONTHS PAYMENT PLAN ($16,860 USD) Buy Now -$2,810 USD
12 MONTHS PAYMENT PLAN ($20,640 USD) Buy Now -$1,720 USD
3 months support package Buy Now -$1,080 USD
6 months support package Buy Now -$1,640 USD
Please DOWNLOAD WebPhone SDK, demo application, or any of our sample codes and feel free to try them out for up to 30 days.
You can develop your SIP based webphone with your own GUI and brand very easily. Sample source code HTML is available on our website, please click the SAMPLES link to find out more.
Yes, video conferencing is supported. Please visit SAMPLES for the trial and testing of video conferencing feature.
Yes, it is really easy to develop such feature by using VaxVoIP WebPhone SDK exported methods. In which supervisor instructs to the agent in real-time. But customer does not hear the voice of the supervisor.
Yes, supported Web Browsers are Microsoft's Internet Explorer, Fire Fox, Google Chrome and other Mozilla based web browsers. Demo webphone sample code is available on the website.
Yes, one can easily develop SIP based Instant Messenger. Please visit the FEATURES link for more details.
Our SIP SDK works without any problem with Asterisk and all other SIP based SERVER and SERVICE providers.
No it does not support Hold feature . Please visit the FEATURES link for more details.
Yes, it is possible and very easy to develop using our WebPhone SIP SDK.
Yes, DTMF generation and detection both are supported. Please visit the FEATURES link for more details.
Yes, we provide customization of our SDK. Please send us your requirements and a sales person will be contacting you soon.
Please click the FEATURES link for more details.
There is no difference, except 'Evaluation Copy' message box and expiry.
No, we don't sell the source code of our SDK.