Telos VXs is a containerized product. It will require some initial configuration from a Telos ProServices Field Application Engineer or one of our worldwide partners. They will assist with deployment of the Docker container for your VXs engine.
When your VXs engine is up and running, log into it using the IP address provided by the container deployment team.
Because VXs is designed to be deployed on-prem or in the cloud, the login procedure is slightly different from other Telos Alliance devices with dedicated hardware. You will need to log in using your TelosCare ID. If you don’t yet have a TelosCare ID, you can register for one by clicking “Click here to register for your TelosCare ID” and filling out a simple form.
Once submitted, you will receive a verification code via the email address you used to register. You’ll need to enter the code to confirm your registration.
After you enter the correct verification code, you will be registered and allowed to log in.
Before provisioning the VXs networks, it’s important to make sure that you have the right kind of network switch for the AoIP interfaces. The switch you choose must be capable of handling multicast audio. If you use a switch that isn’t capable of properly routing multicast traffic (most unmanaged consumer grade switches) you will send all AoIP traffic to all ports on the switch, flooding your network, diminishing performance, and likely interrupting or crashing other devices that might be plugged into your network.
For help with selecting a suitable switch, please see this TelosHelp document.
During deployment of the VXs container, your Telos ProService Field Application Engineer will have configured one or more network interfaces and set their IP addresses. You will now select which interface(s) handle each function - SIP or AoIP.
To set your AoIP interfaces, select the MAIN tab at the left of your browser window, and select at least one interface for use with Livewire+ or AES67 audio via the dropdown menus. In this example, our AoIP network is eno2 and has an IP address of 192.168.2.20. You will see the NIC’s MAC address after the IP address.
If needed, you can configure a second AoIP interface to enable SMPTE 2022-7 redundancy. If you don't have SMPTE 2022-7 requirements, do not specify a NIC for Interface B.
After making your selections, press the SAVE button in the Network Interface area. Confirm your selection in the box that appears, then press the RESTART VXS button in the System Control part of the screen to ensure that all network changes have been applied. Once VXs restarts, you’ll need to log back in using your TelosCare credentials.
When you’ve logged back in, select the SIP tab and choose a network interface for SIP traffic from the dropdown menu. In this example, our SIP network is eno1 and has an IP address of 192.168.1.20.
After making your selection, press SAVE and reboot VXs once again by going to MAIN and pressing the RESTART VXS button under System Control.
While it is most common for VSets to be connected to one of the AoIP networks defined on the MAIN tab, they can also be connected to the VoIP network defined on the SIP tab.
There are separate manuals covering VSet configuration. You can find them on our website here: https://www.telosalliance.com/Telos/Telos-VSet
For a list of all ports and protocols used by VXs, please visit this TelosHelp document.
As of this writing, VXs supports the G.711 (alaw or ulaw) and G.722 codecs. G.711 is the codec used by the PSTN and features bandwidth of 300-3400 Hz.
G.722 is the same codec known to broadcasters from the pre-MPEG days of ISDN remotes. It has a wider bandwidth of 50-7000 Hz, offering much better than usual speech quality.
Not all VoIP providers support G.722, and those that do are limited to using it for calls internal to their own network, as the PSTN does not support G.722. G.722 can be useful in some circumstances though. For example, G.722 calls may be made from a mobile softphone client on the same provider as your VXs system to allow for easy remote broadcasts with better than PSTN quality audio from virtually any smartphone.
SIP always negotiates a codec supported by both ends, dropping to G.711 if a better option isn’t available at both ends of a call.
The SIP configuration page shows the global SIP settings, along with a list of all SIP servers that your VXs is configured to use. You can configure one or many SIP servers.
The Use SRV Lookups feature is not widely used and should be left turned off unless Telos support asks you to enable it.
Port is the TCP or UDP port used for SIP signaling between VXs and your PBX, or hosted provider, or gateway. Leave it at the default of 5060 unless a non-standard port is required by your provider.
Add your first SIP server by clicking the ADD button in the servers section of the page.
The SIP Server field is where you enter the IP address or URL of the SIP server. In this example, the SIP server is a local FreePBX server with an IP address of 192.168.1.11.
If desired, you can add a more descriptive name for each SIP server using the Name field. In this example, the name we’ve entered is FreePBX. (If left blank, the name field will automatically populate with whatever value you entered into the SIP Server field.)
The Outbound Proxy field is for use when your SIP provider has specified an outbound proxy IP address for you to use for your service. This will not be required for all SIP server configurations. More info is available via this TelosHelp document.
The External IP field is available if port forwarding is needed at a router or gateway. If port forwarding is required at the NAT router of the SIP network, this is where the public facing IP address of the router should be entered.
If your server is part of a Local Domain, enter that here.
As you can see from the example, it’s possible that one or more fields will be left empty.
After populating all the necessary fields, press SAVE.
Press the ADD button to add your first SIP extension.
In our example, we’re registering extensions to a FreePBX server. Both Extension and Auth User are the extension number configured in FreePBX. We’ve toggled the Register switch to ON and left the Expires field blank. For Auth Password, we copied the “Secret” field from each extension’s configuration page in FreePBX.
As you can see, not all fields are populated in our example configuration. This is OK.
We recommend configuring SIP extensions as endpoints that use registration. This makes network troubleshooting easier and tests the entire IP connection to the SIP server. Be sure to toggle the Register switch to ON. This will make VXs register with the PBX, essentially logging in, whenever a show referencing that extension is made active.
The Expires field lets you change the interval (in seconds) that VXs will refresh registration for a particular extension. As a general rule, don’t populate the Expires field unless Telos support recommends it to solve a specific problem.
As SIP messages list IP addresses and ports used to transmit audio on (via RTP) it doesn’t work well if the client is in a private LAN but needs to communicate with a SIP provider outside of the LAN. As messages pass through the router, it translates addresses in IP headers, but not the SIP message itself, giving the provider wrong connection info.
Many SIP providers use clever hacks to work around this limitation without any additional support from the client. If you are connecting VXs to a SIP provider that doesn’t provide such a service, don’t worry - VXs has basic NAT support built-in. Contact Telos support for details.
Before going any further in your configuration, you will need to add a license to your system. Copy and paste the license code supplied by your dealer into the License field on the Licenses page and press the ADD button. You should then see the license key you activated, as well as what features it enables in the Licenses section of the page.
If you are unable to activate your license, or if the license count is not correct, please contact Telos support.
The Studios page lists all the studios that are configured for your VXs. It lets you add new ones, and lists the show that each studio is using.
To add a new show, press the ADD button.
The Studio Name field will populate automatically with a suggested name for this studio. You can leave it at the default or change the name to whatever meets your needs.
You’ll then need to set a number of lines that are available to this show. Typically 6 for a VSet 6 and 12 for a VSet 12, but it can be any number, even beyond the capacity of the call controller you will be using with this studio.
For example, if you have a VSet 6, you could configure a show with more than 6 lines, as long as lines 7 and above were configured as fixed lines with auto answer fixed lines enabled. This could be useful if you wish to have an extension dedicated to IFB or intercom, or configured as a listen line. (Note: You would not have control over this line to force a disconnect, and would depend on the calling party to terminate the call.)
Keep in mind that lines are a licensed feature, and the number of lines assigned to each studio counts against the number of line licenses you’ve purchased.
Next, you need to add either Fixed or Selectable channels.
Keep in mind that fixed Channels have a 1-to-1 relationship with a particular line in a show, while selectable channels are allowed to choose any line that isn’t tied to a fixed channel. Selectable channels function the same way previous generations of Telos talkshow systems always have.
To add a fixed or selectable channel, press the ADD button in either the Fixed Channels or Selectable Channels area.
The Display Name field shows how this channel will be advertised alphanumerically on your Livewire network. (This is the same info you would enter into the Source Name field on an xNode.) A default value will populate when you add the channel, but you can change it to whatever you like.
The Output field is where you define which Livewire channel number or AES67 multicast address you want this channel to have on your AoIP network. The adjoining drop down menu allows you to select between 1 ms AES67 Multicast streams or 5 ms Standard Stereo streams. This is the caller audio.
The Manual Backfeed / Input field defines the input audio path from the studio. This is the audio that will be sent to the caller. Depending on your selections, this field will populate in various ways.
If you set the Output field to a Livewire channel number and select Standard Stereo, it will default to an auto backfeed, useful for interfacing with Axia consoles. The Manual Backfeed toggle switch will remain off, and the field will display “auto.”
If you set the Output field to a Livewire channel number and select AES67 Multicast, it will create an auto backfeed for you, which will also work with an Axia console, however, the Manual Backfeed switch will automatically toggle to on, and the field will be populated the information for a Livewire backfeed automatically. If you are using VXs with a phone type source on an Axia console, there is no need to make any changes.
If you set the Output field to an AES67 multicast address, the Manual Backfeed switch will toggle to on, but the Input field will be blank, and will require configuration.
Anytime you click inside the Input field, this box will appear, allowing you to configure everything as needed for your application.
If you are utilizing xNodes to interface to another brand of console, click the dropdown box that currently says To Source and select From Source instead. Enter the Livewire channel number you wish to receive in the Address field and press SET. In this example, we are taking Livewire channel 5302 as the send to caller audio for this selectable channel
The same principle applies if you are configuring VXs for use in an AES67 environment, only you won’t see the To Source / From Source options. Simply put the appropriate AES67 multicast address in the Address field and make the appropriate selections indicating the number of channels in this AES67 stream and which channel to use.
Program On Hold is the audio a caller will hear when placed on hold. Configuration for this works the same way as Manual Backfeed / Input configuration. In most situations, the program on hold audio is fed from the console’s main program buss. This audio should normally be pre-delay to allow callers to interact with hosts in a natural way when their call is taken to air.
Note: You must configure a Program on Hold feed for each studio. It could be a generic program audio or air feed, but if no working channel is assigned, calls on hold may drop after 30-60 seconds. This is because some PBXs, trunks, and some endpoints will hear silence and disconnect, thinking the call was lost.
AEC is an optional, licensed feature which helps when you monitor calls via a loudspeaker in the same room as the microphone feeding the phone. Without a canceller, the caller’s own voice would be sent back to them as an annoying echo.
Prior to configuration, you will need to toggle the Enabled switch to on.
The canceller needs two inputs and produces one output. The Mic Input is fed from the studio microphone. This can also be the entire mix-minus signal to include multiple microphones.
The Reference (CRMON) input is the audio that needs to be canceled. This is the audio that is going to the monitor or preview loudspeaker that caller audio is played out of.
The Output (Backfeed) of the canceler goes to the VXs phone feed input(s.)
As with fixed and selectable channel configuration, these can be either Livewire channels or AES67 multicast addresses.
VXs supports GPIO (General Purpose Input/Output.) It is a useful way to control VXs functions, or get indications of VXs conditions or modes.
Electrical connections are made via Livewire GPIO xNodes. You can also use Livewire GPIO to signal several different commercially available Livewire enabled studio notification systems, which display GPIO indications on a video display, or create your own using Pathfinder Core Pro.
For GPIO Actions, press the ADD button, select the specific action you wish to perform from the dropdown menu, specify a Livewire Channel number and Pin. Since GPIO Actions are inputs, leave Type at the default setting - From GPIO.
The available actions are:
Take next call
Take next ringing line
Hold all calls
Drop all calls
Enable Block All
Disable Block All
Toggle Block All
Toggle Auto Answer & Hold
Mute Ringer
For GPIO Indications, press the ADD button, select the specific indication (labeled Action here as well) then assign a Livewire Channel number and Pin. Since GPIO Indications are outputs, select To GPIO as the type.
The available indications are:
Next call available
Line ringing
Line ringing (Busy All)
Line ringing (non-Busy All)
Call can be held
Call can be dropped
Block All enabled
Auto Answer & Hold enabled
Ringer muted
Delay Dump
VSet phones have access to an address book stored in VXs. There are actually two sets of address books - one tied to each studio, and one tied to each show.
To edit the address book for a particular studio, go to STUDIOS -> ADDRESS BOOK then press the EDIT button associated with the Studio Name you wish to edit. Once inside, type a name to be displayed in the address book in the Name field, then the number or SIP address you wish to store in the Number / SIP Address field. Finish by pressing the SAVE button.
The procedure for editing the address book for a particular show is very similar to the one used for studios. Go to SHOWS -> ADDRESS BOOK and perform the same actions you would in editing a studio address book.
If a number is added to both the studio and show address books, that number will appear in the address book twice whenever that particular combination of studio and show are selected.
In VXs, a show is a collection of lines that are available to a studio. A studio can log out of one show and into another, bringing up a different set of lines on the VSets and other controllers. In the main shows page, you will see a list of shows that are configured and the studios they are currently assigned to.
To add a new show, press the ADD button.
On the show configuration page, we’ve set up a show with 12 lines, 8 of which are labeled Listener Line, 3 of which are labeled Warmline, and one is labeled Hotline.
The Extension and Server fields reference the SIP server and the extensions configured earlier. You can assign the same extension to multiple buttons as we’ve done here. Calls will ring in on the first available line, acting like a hunt group.
The Channel field allows you to designate whether each line will be Selectable (a number of lines switched to a single or small set of faders) or if it will be tied to a particular Fixed channel (a one-to-one pairing of lines and faders.) The order of the fixed channels in the dropdown menu references the order in which they’re listed in the Studio profile.
The Block All toggle specifies whether a particular line is busied when the Block All button is pressed on a VSet.
Finally, the Ringer dropdown can be used if you have uploaded custom ringer tones to the system. This feature is documented later in the manual.
It’s important to note that shows cannot be edited or deleted while assigned to a studio. You will see a notification like this one if you attempt to edit a show that is in use by a studio.
If you see such a notice, go into that studio’s configuration page and deselect it using the Change Show dropdown menu.
VXs has dynamics processing on both the send (from studio to caller) and receive (from caller to studio) audio paths. While we explain how these controls work, the vast majority of users find the default settings work very well and do not need to change them.
The send (to caller) audio processing is fixed and consists of a protection limiter and some EQ. The purpose of the send limiter is to protect the caller from clipping distortion.
The receive (from caller) processing is adjustable and includes ducking level, an AGC, noise gate, and dynamic EQ.
Receive AGC - This helps level calls out, to make each caller sound similar to the last. We recommend leaving this control at 16, which provides the most consistent levels.
Noise Gate - This removes background noise by gating (turning off) the output when audio drops below a certain threshold. We recommend leaving this setting off because it could mistake a soft-spoken caller for noise and gate unexpectedly. If most of the callers to your VXs are professional announcers calling from noisy environments, you may find this feature helpful.
Receive EQ Mode and Additional EQ - This control changes how the EQ algorithm functions. Setting the control to off bypasses all equalization of the caller’s voice. Setting the mode to Fixed will allow VXs to only apply the EQ specified in the Additional High and Low EQ fields. In Adaptive mode, the VX will dynamically adjust EQ levels for callers in an attempt to make all callers sound similar. The Additional High and Low EQ fields will be added to this automatic EQ.
Caller Ducking Level - This control reduces caller level when audio is being sent to the caller. This helps with intelligibility of host voices when the caller and host are speaking simultaneously, and during contentious interactions, the higher the setting the more the host can “shout down” a caller without anyone having to ride gain on the caller’s fader.
SIP signaling is via digital messages, not tones in the audio like it used to be. Like most of today’s VoIP devices, VXs has sounds loaded into it to mimic the sounds everyone was used to with the PSTN.
You can customize these sounds if desired, either disabling them, or by uploading new sound files to replace them.
If you wish to customize any of these tones, the files you upload must be in the following format:
AU file extension
Linear PCM
8, 16, 24 or 32 bit
Ringtones MUST be 8 kHz, 16 bit, mono with 48 kHz sampling.
The easiest way to get audio files into this format is with the open source audio editor Audacity. Open your file, then export it using these settings:
On the File menu, choose Export, then Export Audio…
Use file type “Other uncompressed files”
Under format options, select AU (Sun/NeXT) with Signed 16-bit PCM encoding.
If you are making ringtones for your VSet phones, convert your audio to 8 kHz before exporting by:
On the Tracks menu, choose Resample…
In the “New sample rate (Hz)” box, select 8000.
The various types of tones loaded into VXs are described below.
Call progress tones:
Dial tone - The sound heard when you pick up the handset before a call is dialed.
Ringback tone - Heard when dialing is complete and the called phone is ringing.
Busy tone - Heard when the called phone is busy.
Reorder tone - A fast busy tone. Signals that there are no call paths available.
Error tone - Usually caused by an incorrectly entered number, but can be from other problems during call setup.
Call disposition tones:
Call answered - A clicking sound played whenever a call is put on air.
Caller hang up - A brief, low tone that is heard whenever a caller on-air disconnects.
Line switch - Mimics the sound of a line button being pressed on an old 1A2 key phone.
Caller alert tone - Sent only to the caller when a call is picked up (not played over the air.) It is a brief ding played at the same point in a call as the white noise bursts played by earlier phone systems to help with line equalization.
Choose the DTMF sub-menu to set these tones. VXs is loaded with the same DTMF tones used traditionally with the PSTN, but they are scrambled on-air so listeners can’t easily decode the number being dialed. (The correct DTMF tones are played to the VSet handset and headset jacks.)
One thing to note about SIP and DTMF. The DTMF tones generated by VXs are merely played for the audience and operators to mimic traditional phone service. It is sometimes necessary to send DTMF to the PSTN after a call is connected, such as for dial-up remote control systems. In this case, VXs sends a SIP message to the PBX or gateway to generate the corresponding DTMF tones. The scrambled DTMF tones used by VXs for privacy do not impact this process at all.
Ringtones play through the VSet speaker when a line is ringing (if not muted via GPIO or VSet settings.) Should you wish to upload custom ringtones, please note the audio file requirements listed above.
These pages provide some insight to the inner workings of VXs for troubleshooting.
The studio information menu provides a list of studios, the number of lines in use by that studio, the number of fixed and selectable channels handled by that studio, and the number of devices (VSets and other controllers) logged into the studio. Pressing the SELECT button for a particular studio will provide more information about that studio.
This page shows the SIP registration status for each line currently in use by the studio.
In this example, we intentionally used an incorrect password for line 5 to show the difference between when a line/extension is registered properly, and when registration fails.
The devices page shows us what devices (VSets, Consoles, PCs and other controllers) are currently connected to VXs. The IP address and ports are listed, as well as the studio that owns the device.
This page displays statistics about the streams handled by the AoIP network interface(s.)
This page displays all of the hardware and health statistics for VXs.
Before using VXs, you will need to add a license code to enable the functionality you have purchased. To add a license, copy the code from the software license certificate provided by your dealer into the License field and press ADD. Afterward, you will see the license code under Licenses, below, as well as the activated features.
If a particular license code has an expiration date, a countdown to expiration will appear. If a license is perpetual, instead of a countdown, the word “Never” will appear in the Expires field.
As of this writing, the Node name and Activation Refresh/Release buttons serve no purpose and can be ignored. They will support future functionality.
The User Management page shows all of the users currently registered to the VXs with their associated TelosCare ID email address, or local account.
Though a TelosCare ID is required for the first login to a VXs after container deployment, you can create one or more local user accounts. To create a "user" account that functions like the user account on (for example) a Telos Alliance xNode, press the ADD button. On the configuration page that appears, leave the Username field set at its default of "user." Press ADD again to create the generic "user" account. You will now be able to log into VXs by clicking LOG IN at the main login page.
To change the password of a user, press EDIT.
To remove access for a user from this VXs, press DELETE.
Click BACKUP SETTINGS to download a backup configuration file. If a login dialog appears, please type user in the user name field, nothing in the password field, and press OK. After doing this, your backup will download as any other file would in your browser.
To restore your settings from a backup, click the RESTORE SETTINGS button, then navigate to and select the backup file on your computer.
When this dialog box appears, press OK to proceed. Any existing settings on your VXs will be erased and replaced with the backup file you loaded. The system will then reboot with the new settings in place.
This tab does exactly what you think it would - it logs you out of VXs. Be aware that you will be logged out instantly. No confirmation dialog box will appear. Should you wish to log back in, you will need to use your TelosCare credentials to do so.
