After downloading the latest version, please go to
the Quick Start Guide to start using the SDK. According
to your needs you can also browse the sample programs
page where you can find source code explanation for the various solutions.
You can address all your questions and requests to us via email at
firstname.lastname@example.org and we will provide
you with all details. It is also possible to chat with one of
our representative by using BeeWebphone. The click to call button of BeeWebphone can
be found on the left side of this page.
There is a great chance for efficient cooperation. All you need to do is to write the
requested extra functions to us (email@example.com).
You will receive the response email soon with our answers and time schedule in case
You can always be up-to-date regarding the latest improvements for the SDK by checking
the changelog on the Download page. There are
new releases of the SDK continuously in order to ensure higher user experience.
To make voice calls you need to connect your system to the telephone network.
This can be done in three ways:
Option #1: You can use a VoIP telephone adapter.
A VoIP telephone adapter is a hardware device that can be connected to your ethernet
LAN or to your computer. It makes it possible to dial telephone calls. There are VoIP
telephone adapters for GSM lines, for standard analog telephone lines and for ISDN
lines. (If you don't have a VoIP telephone adapter, you can order one directly from us).
When you put this hardware on your Ethernet LAN, it will receive an IP address.
You need to configure this IP address in Ozeki VoIP SIP SDK to be able to make voice
telephone calls through the device.
Option #2: You can use a SIP Account provided by Internet based VoIP telephone provider.
VoIP telephone service providers offer a telephone service over the Internet. They can
forward calls to traditional telephone networks and to Internet users signed up to
their network. There are many VoIP telephone service providers out there. When you
sign up for their service, you will receive a SIP account (an IP address, a username
and a password, plus some other details). To make voice calls, you need to
configure the SIP account details in Ozeki VoIP SIP SDK.
Option #3: You can use your existing office PBX if it is a VoIP system.
If your office has an IP telephone system, you can connect Ozeki VoIP SIP SDK to it
through the office LAN. Ozeki VoIP SIP SDK can log in with a SIP account to the PBX
and it can make telephone calls just like any other office desktop phone would.
This is a good choice if you have already invested in IP telephony, because in
this case you don't need to have a dedicated telephone line for outbound voice
telephone calls made with Ozeki VoIP SIP SDK.
You can use the software in trial mode for a period of 20 days. If this time has
expired, you need to purchase a licensed software in order to use it for unlimited
time. When you purchase Ozeki VoIP SIP SDK you will receive a registration code
which helps you to activate the software.
Ozeki VoIP SIP SDK can be used to create a C# softphone or a VB.net softphone in
minutes or you can build SIP VoIP call services easily and quickly. After download
you can customize the SDK to provide seamless and efficient VoIP services. Without
spending time with learning SIP basics you can instantly start your project and
boost the benefits of VoIP technology in your application.
It occurs because the SDK is behind double NAT while the PBX and the 3CX phone are
behind simple NAT. When the SDK registers to the PBX, it ‘notices’ that they are
not on the same network. That is why, it turns to the STUN server. Since the STUN
server is on public net, it returns the public IP address of the client to the SDK,
this way, the SDK forwards this public IP address to the PBX.
To overcome this issue, please try one of the following options:
The network connection of the virtual machine should be bridged instead of
NAT (This is the simpler option).
Install a STUN server to the network on which the PBX is installed, and set
this STUN server for the SDK.
If the sound is played slower, that seems like a convert problem. For example:
A 44kHz sound is played on 8 kHz, the result will be a slow and lengthen sound.
This can happen when the microphone records sound in 22 kHz format but the PCMU
codec (for the call) uses 8 kHz. To solve the issue out, it is necessary for the
codec to convert the 22 kHz sound data into 8 kHz sound data.
If you use MediaConnector and PhoneCallMediaSender classes (the SDK includes them),
then this procedure is automatic.
Please check if all PBX registration credentials are correct in the WPF example.
Try to change the NAT settings from:
sorbansoftPhone.ChangeNATSettings(NATTraversalMethodType.ICE, "", "", "");
softPhone.ChangeNATSettings(NATTraversalMethodType.NONE, "", "", "");
Also, a certain port range should be opened on the firewall for the RTP packages to come through.
Yes, your idea can be realized with Ozeki VoIP SIP SDK efficiently.
You only need to hook up to the event 'MediaDataReceived' in the PhoneCall class
of the SDK. (MediaDataReceived is the event of receiving an incoming audio packet.)
If the DTMF signal comes via the RTP stream or via SIP INFO method then the saved
conversation won't contain the DTMF sound.
But if the DTMF signal is mixed into the audio stream then filtering is not possible.
(e.g. in case of an analog system or if a land-line phone is on the receiver's end)
When the SDK talks with various PBX systems, it uses the SIP protocol for signaling and
the RTP protocol for sending/receiving audio. In the RTP stream, the audio is tansferred
using VoIP codecs, such as G711, G722, etc. These codecs are supported by all VoIP PBX
systems (all brands).
When you play a wav file into a phone call, the wav file is first decoded to raw audio,
then it is encoded according to the codec required by the RTP connection. In other words
wav is the input for the SDK and a G711 encoded RTP stream is the output.
Ozeki VoIP SIP SDK supports uncompressed PCM wav files for input. It converts these wav
files to the appropriate format when it talks with VoIP PBX systems, VoIP service providers
or VoIP telephones.
So if you use plain PCM wav as input it will be compatible with all brands.