CHAPTER 2 |
8
INSTALLATION AND CONFIGURATION
2
Installation and Configuration
The diagram shows a generic installation that could be used in a typical studio. Two Ethernet ports are employed on the VX
engine to isolate the two networks that are present in this example. L audio network is handled by the LAN port,
while SIP and/or Internet provided network is handled by the WAN port. Since it’s handling L or AES67, the switch
connected to the LAN port must be configured for multicast audio. Meanwhile, the switch connected to the WAN port can be a
switch on your regular business network.
Telco lines enter two ways: From the PSTN via a PBX or from a SIP VoIP provider over an IP network, configured as endpoints.
In most PBXs these are called generic or third-party SIP extensions. xNodes provide analog, AES, and/or GPIO connections
to traditional studio equipment. The audio xNode also creates the clocking signal for the L network. xNodes are
not needed when the studio equipment can accept AoIP natively and if there is already a clock source on the network. For an
example of this sort of installation, see Axia Console as VX Controller in Section 5.
Connecting to the PSTN: PBXs and Providers
‘PSTN’ is an acronym for ‘Public Switched Telephone Network’, the traditional telephone network is now made up of many
types of service providers, including the Legacy Utility Telco providers often called Local Exchange Carriers (or LECs),
Metro Ethernet providers (availability varies regionally), Fixed Wireless providers, Internet Telephone Service Providers
(who deliver service via the public Internet) and Competitive Local Exchange Carriers (CLECs). Most providers and PBX
manufacturers have adopted support for basic SIP trunks and also some provider services provisioned as “Endpoints”. They
also usually offer traditional interfaces like POTS, ISDN (PRI), often delivered over Ethernet or T1’s/E1’s as ‘last-mile’
connection technologies.
The VX system connects to Telco lines using industry standard SIP (Session Initiation Protocol). This means it is compatible
with a wide variety of VoIP services, gateways, and PBXs. Gateways can be used to interface legacy PSTN lines to SIP. We
discourage this as we’ve found them difficult to set up and maintain. They simply don’t provide the flexibility, audio quality
and reliability of an all-digital connection. There are other small drawbacks that wind up being surprises down the road like
when your callers receive a ring back signal instead of a busy signal when your lines are in pre-contest Block All mode. This
happens because there is no ‘off-hook’ condition in the SIP world.
