In this video, we’re going to continue expanding on the VIRTUAL view of AVX Architect and look at some basic Voip properties. We won’t cover all the available VoIP settings in this video, but we will point to some of the most important settings for getting started. VoIP is available on both the Omni 256p and 512p processors. The only difference between their VoIP capabilities is the available channel count.  The Omni 256p has 32 channels of VoIP while the 512p has 64 channels.
Let’s start with the settings available in the SYSTEM view. Start by selecting the processor in the workspace. In the PROPERTIES section of the right pane, you’ll find a section for VOIP. One quick note about one setting above the VOIP section, the Network Traffic Mode. This setting is important because by default, VoIP will be set to be on the secondary HControl port. This can be changed to allow VoIP on both ports in case of daisy-chaining processors, or to isolate VoIP and HControl. Check out our video on Network Traffic Modes for more information. Back in the VOIP properties, most of these settings don’t need to be changed, and that’s why we’ll cover them in greater detail in a different video. Here, you will find the Domain name, or IP address of the VoIP domain. Proxy address if you’re using a proxy. Transport Type, if you want to manually set the transport type between TCP, UDP, or TLS. Last is the VoIP CPU Configuration, which will allow you to change the VoIP interface between DHCP and a static address. 
Let’s look at VoIP in the Virtual View now. The VoIP processing objects can be found in the Conferencing section of the Audio processing objects. This will give you both a VoIP input and output object. In the PROPERTIES window, you’ll find the usual naming, workspace location, and DSP allocation settings. The Options section will allow you to change various tones from the default tone. End Tone is the tone that plays when a call ends. Ring tone is the sound of the ringer. Answer tone is the tone that plays when a call is connected. Switching these from the default tone to a custom tone will add an input node to the output block that can be fed with a tone generator or media player object. 
The DTMF mode can be changed under the tones. The default mode is Out of Band, but In Band, SipInfo, or None are also options. The DTMF can also be set to have a fixed length with defined on and off times. Setting the fixed length to Off will cause the DTMF to play as long as a button is pressed. The Codec setting is Automatic by default, which will allow the processor to try and detect the best codec to use for a call, but this can be changed to a specific codec. Finally, we have User Name, Authentication Name, and Password for this line, as well as a Display Name, which is the name displayed when you call or dial out.