Let’s define what “uses VoIP” means. The VX uses it in two distinct ways: One, it can connect to Telco services using standard SIP VoIP. You benefit from having options—connecting to SIP providers and SIP PBXs digitally, or to ISDN and analog lines via gateways, Asterisk servers, etc. With VX you can finally integrate your on-air phones with office phone systems from a variety of vendors. Getting Telco service from VoIP dial-tone providers means that your audio quality and hybrid null will be much better as VoIP dial tone is delivered “4 wire” without hum, noise, and loop loss. Building on ubiquitous VoIP standards means a variety of third-party hardware can offer flexibility, and you might save a lot of money getting service this way. Two, VX system components connect to each other over standard IP/Ethernet networks with all the advantages that brings. For example, in Livewire-equipped facilities, one RJ-45 jack connects dozens of audio channels and rich control to phone-like controllers, PC applications, integrated console controllers, etc.
VX Prime is a smaller version of our popular SIP-based VX Broadcast Phone system. VX Prime follows the same architecture and philosophy as VX, but in a system that is limited to just eight telephone hybrids (or console faders).
Many customers told us they really liked VX, but the cost of a full VX system was outside their budget and the capabilities were far greater than they would ever need. They wished for a similar system with reduced capacity at a lower price point. VX Prime fits that need with a hybrid count that is one third of a full VX system, at a price that is 40% lower.
All VX systems let you leverage inexpensive networking to serve your entire facility. Because of this, cost-per-studio is surprisingly reasonable. You’ll use VX in your on-air studios to replace older multi-line systems, and you’ll use it to replace hybrids in newsrooms and production studios. You might also decide to eliminate walls full of “couplers” for pre-delay IFB dial-up lines, and transitioning from Telco POTS lines to SIP-based services will also save you a lot of money every month. We’ve seen stations saving thousands of dollars a month (no kidding) by eliminating POTS and ISDN lines, with all their extra taxes and fees. Add to that that VX Prime is about 40% less expensive than a full VX system, and you’ve got a recipe for big studio sound, at small studio prices.
Currently, VX Prime does not include the Acoustic Echo Cancellation (AEC) technology of the VX, but your calls will still sound great because VX Prime has the same caller AGC and Digital Dynamic EQ (DDEQ) as the VX. AEC will be a no cost upgrade in the near future.
In general, stations needing to put phones in two to four studios will find that a VX Prime meets their needs perfectly. And the cost will be significantly less than buying several multiline phone systems to distribute amongst studios. Stations with five or more studios (or fewer studios but with heavy phone usage) may be able to squeeze into a VX Prime, but should consider their needs carefully to make sure the full VX is not a better match.
No, VX Prime is a fixed solution with no capability to increase the fader/hybrid count. The design philosophy of VX Prime is to give stations that would never need the capacity of a full VX a product that will fit their needs and budget.
You could say that the theoretical maximum is 96 (eight studios of 12 numbers or line appearances each). However, there is no limit to the number of shows that you can have and you can only have one show at a time in use in each of up to eight studios. With modern digital systems, there are really no "lines" in the classic sense. A POTS line has a single telephone number for a single physical pair of wires. A digital system (SIP or ISDN) uses Direct Inward Dial (DID) telephone numbers to route the call to the correct user’s phone over a free channel. However, to a user, this all feels and works exactly like a familiar old time "line."
Sure, since the VX Prime is an SIP-based system, there is no restriction to the number of SIP lines or phone numbers that can come into the system. The only restrictions are in the number of SIP lines that can appear in a given “show” definition (12 for each show), and the number of callers on the air simultaneously across all studios (8 callers actually on-air).
Yes. If you are a Livewire plant, then congratulations on a great choice that just saved you some additional money and simplified the integration of VX Prime into your facility. This is because VX Prime will connect to your existing Livewire network with a single Ethernet cable and provide phone hybrids into each of your studios without need for any additional wiring or physical audio connections. If you are not a Livewire plant yet, then you can use our convenient Telos Multipurpose Node audio interface to provide audio and GPIO connectivity to your studio consoles. One Telos Multipurpose Node is usually just right for each studio.
We suggest that you request a short call with a Telos support engineer before an order is placed, who can run through a quick checklist to determine your needs and make suggestions that can save you money and help you plan your installation. There is no cost for this, and we do it for most new systems. Additional on-site provisioning and training is available for a modest fee. Telos Support can be reached any time at +1 216 622 0247.
You can do that, and it’s not difficult. You only need an Asterisk server or POTS gateway device. However, we encourage using direct digital delivery for the best sound quality – and SIP VoIP is almost always significantly less expensive than POTS.
Same as with POTS: If you have ISDN now and want to keep it, there are gateways available. However, it is often cheaper to port these numbers to a VoIP dial tone provider. All of these options are worth considering.
Call us! Telos experts will help you select and configure an Asterisk system to suit your station’s needs.
Since VoIP depends on SIP, and SIP is a defined standard, the only question is if your PBX can provide a trunk or endpoint extensions between itself and your VX system. If so, your PBX will work with VX—but if you have any doubts, call Telos Support and we’ll help you figure it out.
If your existing PBX cannot provide an SIP trunk line, it may be too old to offer such service, or might need upgrading. Contact your PBX provider and ask if your PBX can be upgraded for use with VoIP systems. Often, we integrate Asterisk servers with older digital PBX’s using PRI trunks between the systems. Digital (4 wire) is always better than POTS! We can help you through this. Contact support and set up a call with a Telos Telephony expert.
Actually, we recommend it! We think that you’ll find that they work better than you may have expected, as many of the VoIP problems we have seen are caused by limitations of analog terminal adaptors (IP to POTS gateway devices)! Since these are not needed with a VoIP-based system such as the VX, that class of problem is eliminated. There are a number of inexpensive ways to try VoIP without risk. You can also get VoIP-delivered numbers from distant area codes and exchanges. If you’re paying mileage for foreign exchange lines or have national toll free numbers, you’ll definitely want to consider this option. Talk with us!
Unlike ISDN, SIP service is actually pretty simple. Your VoIP provider will give you the IP address and registration information you will need. Once you have this, enter it into your VX Engine using the SIP Configuration web page in the VX Control Center. The VX uses what the industry calls “endpoints” rather than trunks. Endpoint is the new “IT Guy” word for phones, while trunks are circuits between switches (central office and PBX in this case).
Sure! AT&T (or any SIP endpoint service provider) will provide you with the authentication info necessary for the connection between VX and your VoIP provider. Historically, the newer Competitive Local Exchange Carriers (CLECs) have been earlier and more proficient with new technologies. including SIP. I can’t put a flasher across a VoIP line, so how can I flash a light when the hotline rings? This same issue arises with ISDN, so since about 1998 we’ve been including a GPIO output for this function. The VX has this capability. In fact, it has multiple outputs which can be assigned to any of your VoIP lines.
We get this concern often and understand why you ask. The term Voice over Internet Protocol (VoIP) does not distinguish between “VoIP over the Internet” versus “Voice over other (managed) IP networks.” If you have been keeping up with the transition to IP Codecs, you probably have noticed the same terminology issue there. IP-based STLs over an IP T1 are just as reliable as traditional STLs over traditional TDM T1 circuits. You are completely right to be concerned about a VoIP trunk (or an STL) over the Internet, as that is not at all the same thing, and performance in that case could be variable. Inside the facility, on a LAN, all problems disappear, since you have plenty of bandwidth and full control over the network.
Well, there’s audio quality. In the early days VoIP used a lot of compression, with bit rates being as low as 6kbps. Needless to say, the resulting audio was not impressive. These low-rate codecs have mostly fallen by the wayside. The lowest-grade codec the VX supports is G.711, the pre-IP standard for digital audio in the telephone network. You will eventually benefit from higher-fidelity codecs as these proliferate in the VoIP world.
Despite the similar names and underlying technologies, they are very different with regard to performance and application. An analog phone line and a balanced 600? studio audio circuit are pretty much the same tech, but the applications and performance are very different. “AoIP” has come to mean professional studio-grade audio networking—full-fidelity and usually with no compression. Low-delay and synchronized channels are other distinguishing characteristics. Another way the two differ is that AoIP uses an advertising/ discovery protocol for receivers to find sources instead of the Session Initiation Protocol (SIP) that VoIP employs. AoIP uses a system-wide clock mechanism to support low-delay and tightly synchronized channels. Finally, AoIP often takes advantage of IP’s multicast capability to permit multiple receivers to listen to an audio source efficiently. AoIP is intended for managed, guaranteed-bandwidth networks, such as with an Ethernet switch as the core of a local area network.
Now you are very close to VoIP! IP codecs use SIP for call setup and various codecs for compression, so are similar to VoIP telephones. In fact, they actually are VoIP telephones and can sometimes interoperate with them. They have better codecs than VoIP phones, though. AAC-ELD, in particular. Advanced IP codecs, such as our Telos Z/IP ONE, employ sophisticated technologies to overcome the Internet’s deficiencies. They include dynamic buffering, error concealment, and other clever stuff.
IAX is a protocol invented by Asterisk. Asterisk is open-source Linux-based PBX software that runs on a PC. The VX /Asterisk PBX are an attractive combo that has become popular within the broadcast industry. IAX provides functions similar to SIP, but with more bandwidth efficiency. VX doesn’t support SIP or IAX trunking at this time. But you can connect the VX o an Asterisk with SIP or IAX trunking or as multiple SIP extensions. There’s plenty of bandwidth on a LAN, so this works fine, while staying with a standards-based approach. We like Asterisk as a VX adjunct. It can add voice mail, automated attendant, blocking callers from caller ID, off-premise SIP extensions, and more to a VX installation, as well as handling the interface between POTS/ISDN/T1 phone lines more efficiently and with less configuration confusion than standalone “gateway” boxes.
A “one-size-fits-all” Asterisk box just isn’t practical due to the variety of configuration options needed to tailor such a system to your station’s specific equipment and Telco service. We’ll be happy to provide resources that can help you get up and running, though.
Yes, we can assist. There are several types of VoIP dial-tone providers. You’ll want to consider how the service will be delivered to you; via the Internet (like Vonage), or via a dedicated IP circuit from the provider that includes a Service Level Agreement and guaranteed Quality of Service (as offered by a number of vendors including most of the traditional Telcos). For discussion of this and other matters, contact Telos directly. Besides audio quality and reliability, when dealing with the internet, there are also important security considerations. Telos Telephony experts can help you with all of these.
As we hinted above, yes! VX systems support G.722 (7khz, ‘wideband audio’) and G.711 (3.4khz, “phone quality”).
Yup. VX systems connect to VoIP on the Telco side and Livewire AoIP on the studio side. This makes integration with Axia consoles and networks easy and efficient. If you don’t already have a Livewire network, you would use Telos audio nodes, like our Telos Multipurpose Node, to provide I/O in Analog or AES/EBU format, as well as GPIO breakouts.
Advanced audio processing, and the fact that you never have to overcome Telco loop losses or extra two-to-four-wire conversions means that the voice quality is as good as it can be. Mobile phone calls can be less than perfect at times, but VX recovers the best possible audio from them. Caller audio is maintained digitally and four-wire end-to-end, and VX does a clean sample-rate conversion from the Telco rate to Livewire’s 48khz. There is also a sophisticated dynamics processing section and automatic EQ designed with help from our colleagues at Omnia.
Yes. Many VoIP phones, for example, support the G.722 codec. VX systems support this, as well. An extension, known as G.722.2., is called “HD Voice.” It’s 7khz and doesn’t sound at all like traditional “phone audio”—in fact, it sounds better than regular G.722! Telos has partnered with Luci, the developers of Luci LIVE and Luci LIVE Lite, so that your field reporters with the Luci app on their smartphones can call directly to VX systems for really phenomenal sounding on-scene remote drops.
Setup is done using any Web browser on a PC or laptop connected to your VX’s network. It may be little different than what you’re used to, but power and flexibility do come with a little complexity. We have a team of VX experts to guide you through the process, and we’re always at your side via 24/7 support should you need us.
We recognize that any time you change anything in a studio, there can be some transition time. While there are a lot of new features in VX, your staff can use the basic features immediately because it works just like familiar and comfortable Telos gear. Operators familiar with our longstanding two-column line selection will be right at home. The color, hi-rez LCDs and seamless console integration for Axia clients enhance the user experience. As you read this, systems around the world are screening calls and putting them on the air with ease.
Yes. You can dedicate a fader to a particular line (We call these “fixed” lines), and you can have as many as you want or you can “lock” VIP callers “on” to prevent dropping them accidentally. Either technique can be used for callers who stay on-air while other callers are coming and going.
This is one of the strengths of VX. Since VoIP calls are already four-wire, multiple lines can be easily conferenced with very high quality. And the user interface lets operators assign selectable lines to single or multiple faders.
Yes. Telos plus Axia is the dynamic duo, the ultimate in studio technology. You start with Axia, the most flexible console/audio platform, then add smoothly integrated phones with the IP network powering it all. The network delivers any of your Telco lines to any of your studios, in any combination. Any line is available in any studio at any time.
VX systems have built-in firewalls to isolate the VoIP connection from your studio network. We use this same approach in all Telos equipment with features that bridge multiple networks.
VX Prime comes with a basic call-screening app, Xscreen2, from the call-screening experts at Broadcast Bionics. Other networked, PC-based apps put information about your callers in front of your producers without the need for caller ID boxes, serial cables, or other hassles.
Using Broadcast Bionics’ PhoneBOX VX, yes! Telos has always built open systems to allow others to create their own visions around our gear.