
OVERVIEW
VaxVoIP SIP Phone 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 software applications. It accelerates the development of SIP softphone having your own GUI (graphical user interface) and brand name.
VaxVoIP SIP SDK can also be used to develop softphone for MS Windows OS, Android OS, Apple iPhone, Apple iPad and other Hand-Held devices.
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.
FEATURES
Add many new and powerful softphone related features and just discover, how much better your applications can be with:
VaxVoIP SIP Phone SDK export funtions to activate SIP based video conference.
VaxVoIP SIP Phone 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 SIP SDK supports Voice Changer, which works in real-time and let you sound like a robot, a chipmunk, a drunk grandpa, a teen boy or someone who just inhaled helium.
VaxVoIP SIP SDK exports functionality to develop interactive intelligence based answering machine detection feature. Please run sample code and demo application for more details. Such feature is only available for MS Windows SDK.
It is really easy to develop softphone for Android based devices (HTC, SAMSUNG, XPERIA etc.), Apple iOS based devices (iPhone, iPad, iPod etc.) and other Hand-Held device. Please download the sample codes and SDK for more details.
VaxVoIP SIP 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 softphone. VaxVoIP 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 sample code and demo for more details.
You can dial and receive phone calls through any SIP based server, gateway or ITSP (Internet Telephony Service Provider).
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.
User can dial and receive multiple calls to start conference call.
During the call session, user can put any line on hold.
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.
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.
VaxVoIP SIP SDK offers advanced Noise Cancellation technology that allows significant suppression of any background noise and produce high quality of output speech.
We support AGC (auto gain controller). AGC is a mechanism by which input voice gain/volume is adjusted automatically based on input signal level.
During the phone call, you will be able to record the conversation into wave (.wav) file for later play back.
VaxVoIP SIP SDK export methods to play wave (.wav) file to the remote end.
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.
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.
VaxVoIP SIP SDK supports for both narrowband and wideband codecs that's why it works with all type of Internet connections.
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 SIP SDK also support DTMF tones detection feature.
VaxVoIP SIP SDK support DND (Do Not Disturb) feature.
User can control Mic and Speakers volume direclty.
Due to the support of both NARROWBAND & WIDEBAND voice codecs, VaxVoIP component works with all kinds of Internet connections.
After purchasing the license key, you will get the one year product new versions and upgrades free of charges.
SAMPLE
MS WINDOWS DEKTOP OS
Visual cSharp .NET
Visual Basic .NET
Visual cSharp .NET (WPF)
Visual Basic .NET (WPF)
Visual C++
Delphi
ANDROID OS
JAVA
APPLE iOS
OBJECTIVE C
SWIFT
As we are happy to provide a 30 day free trial, please download any sample code and try it out.
DEMO
MS WINDOWS DESKTOP OS
Softphone - Desktop Application Download APP
ANDROID OS
Softphone - Android Device Available at Google Play Store
APPLE iOS
Softphone - Device iPhone/iPad Available at Apple iStore
DOWNLOAD
LIBRARY FOR ANDROID OS (.SO)
VaxVoIP SIP Library (.so) allows to develop SIP based VoIP Softphone for Android OS. It is develped in Android NDK and can be used in Android Studio based software projects. Please download the sample code for more details.
LIBRARY FOR iOS (.A)
VaxVoIP SIP Static Library (.A) for iOS is the easiest way to develop softphone for Apple iOS based iPhone, iPad and iPod devices.
It is developed by using ObjectiveC++, Cocoa Library and other frameworks. Please download (ObjectiveC++ or Swift) sample code and open it using latest version of Xcode and have a look for more details.
COM COMPONENT FOR DESKTOP PC (.DLL)
VaxVoIP SIP COM component (.dll) is the best way to incorporate SIP features in your Delphi, Visual C# or Visual Basic .NET applications. COM component should be registered first before using its exported methods. To register, the COMM dll 'regsvr32' utility can be used. For example; regsvr32 VaxSIPUserAgentCOM.dll
For more detail, Sample code for Visual Basic .NET, Visual C# and Delphi are available on the website.
LIBRARY FOR DESKTOP PC (.LIB & .DLL)
VaxVoIP SIP Library (.LIB) is suitable to incorporate SIP features in your Visual C++ based applications. For more detail, Sample code for Visual C++ can be downloaded from the website.
PRICING
We offer single product/software based license to use VaxVoIP SIP SDK. You are only limited to use one license with one of your product/software.
Pay one-time cost to buy the VaxVoIP SIP SDK license, use it in your desktop/iOS/android application and you are free to distribute your application to any number of your customers without paying extra charges. For more details, please contact us.
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.
Basic Buy Now -$2,640 USD
Audio
Chat
Video
Others
Upgrade to Standard Upgrade Now -$1,480 USD
Upgrade to advance Upgrade Now -$3,240 USD
Basic Buy Now -$2,620 USD
Audio
Chat
Video
Others
Upgrade to Standard Upgrade Now -$1,480 USD
Upgrade to advance Upgrade Now -$3,210 USD
Basic Buy Now -$2,480 USD
Audio
Chat
Video
Others
Upgrade to Standard Upgrade Now -$1,468 USD
Upgrade to advance Upgrade Now -$3,240 USD
SUPPORT PACKAGES
3 months support package Buy Now -$1,060 USD
6 months support package Buy Now -$1,680 USD
FAQS
Please DOWNLOAD 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 softphone with your own GUI and brand very easily. Sample source codes for VB, VB.Net, C#, Delphi and VC++ are 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, Voice Changer is supported, which works in real-time and let you sound like a robot, a chipmunk, a drunk grandpa, a teen boy or someone who just inhaled helium.
Yes, answering machine detection with interactive intelligence is supported. For testing, please download the sample code or demo application and run it.
Yes, it is really easy to develop such feature by using VaxVoIP SDK exported methods. In which supervisor instructs to the agent in real-time. But customer does not hear the voice of the supervisor.
Yes, it is really easy to develop softphone for iPhone, iPad and other android based phone and tablet devices. Please have a look at demo and sample codes available on the website.
Yes, one can easily develop SIP based Instant Messenger. Please visit the FEATURES link for more details.
Yes, it works without any problem on both 64bit and 32bit versions of MS Windows OS.
Yes, it is supported, 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.
G711 A-Law, G711 U-Law, G.729, iLBC and GSM 6.10 are supported.
Please visit the FEATURES link for more details.
Yes, it is supported. Please visit the FEATURES link for more details.
Yes, it is supported. Please visit the FEATURES link for more details.
Yes, it is supported. Please visit the FEATURES link for more details.
Yes, phone conversation recording feature is supported. Please visit the FEATURES link to find out more.
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 FEATURESlink 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.