If you want to change the IP address of a VSet phone, press the Menu button and then navigate to the Setup Menu. Pressing the button to the left of the Setup Menu once will show you the current IP address: For a 12 line VSet, press #8. For a 6 line VSet, press #6 to get to next screen, then #4. To change the IP address after navigating to Setup, press and hold the button to the left of the Setup button for 3-5 seconds, then release it. Once you release the button, the IP address will be shown and editable. Should you need to correct an entry error, use the large X assigned to one of the 8 soft keys on the right side of the LCD display.
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 PBXs digitally, to ISDN and analog lines via gateways, 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 phones-like controllers, PC applications, integrated console controllers, etc.
Despite the similar names and underlying technologies, they are very different with regard to performance and application. An analog phone line and a balanced 600Ω 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! These 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. AACELD, in particular. Advanced IP codecs, such as our Z/IP, employ sophisticated technologies to overcome the Internet’s deficiencies. Dynamic buffering, error concealment and more clever stuff.
Yes! The VX supports g.722 (7khz, ‘wideband audio’ and g.711 (3.4khz, “phone quality”).
We know that engineers are lifetime learners and encourage that. However, just as you probably don’t know much about “SS7” or “IUP” in the telephone network, understanding these details is optional. We do have White Papers on our web site to educate you on these, and other, terms. You could also read Steve and Skip’s AoIP book, too, for a fun and comprehensive coverage of this stuff.
A basic Call Screening app, VX Producer, comes with the system. Other networked, PC-based apps, such as Broadcast Bionics’ PhoneBOX VX or NeoGroupe’s applications 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 Phone Box VX, yes! Telos has always built open systems to allow others to create their own visions around our gear.
VX Producer runs on Windows PCs, so it uses the default sound device you've chosen in the Windows Control Panel. You may change that setting to use any other Windows sound device that you prefer - we recommend a standard US mic/headset combo. No other driver or interface is needed.
VX uses the Livewire AoIP standard to route audio, and we have a list of recommended switches. Please see Axia for a current list of supported switches.
The VX has enough processing horsepower to deal with even extreme echo situations, and four-wire Telco delivery means that the only external echo path is from the caller’s line, when it’s analog, and the caller’s handset. The VX uses Telos’ latest hybrid technology (5th generation), enhanced with the latest state of the art acoustic echo cancellation. Even when using open speakers, and changing levels during a call, the new algorithm makes feedback nearly impossible.
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. Calls from mobile phone calls will be less than perfect at times, but VX extracts the best possible from them. Caller audio is maintained digital 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.
Yup. The VX takes 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 Axia analog or AES audio ‘nodes’ to provide I/O in either format. Each node provides eight stereo inputs and eight stereo outputs to and from the system. Each Axia GPIO node provides 8 “groups” of 5 inputs and 5 outputs, covering the needs of 8 studios. Telos Support is always available to help you specify exactly what you need.
Yes, we can assist. We have been working closely with the VX beta sites and other early adopters of VoIP, so we have plenty of experience to share. There are several types of VoIP dialtone 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, check the Telos web site on a regular basis as we continue post material on this and related topics.
This is a matter of taste, but a general guideline is that analog lines are best interfaced with an external gateway to optimize audio quality, while ISDN BRI and PRI can use Asterisk cards to have a lower-cost solution.
A pre-configured 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.
IAX is a protocol invented by the Asterisk people. It provides functions similar to SIP, but with more bandwidth efficiency. The VX doesn’t support IAX trunking at this time. But you can connect the VX to an Asterisk with SIP 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. Asterisk is free Linux-based PBX software that runs on a PC. The VX and Asterisk PBX are an attractive combo we expect will become popular within the broadcast industry.
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.
This same issue arises with ISDN, so beginning with our TWOx12 we included a GPIO output for this function. Not to worry – The VX has this capability. In fact, it has multiple outputs which can be assigned to any of your VoIP lines.
Sure! AT&T (or any SIP Trunking provider) will provide you with the authentication info necessary for the connection between VX and your VoIP provider.
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 standard for digital audio in the telephone network pre-IP. And you will eventually benefit from higher-fidelity codecs as these proliferate in the VoIP world.
Setup is via web. It may be little different than what you’re used to (or not) but it’s not difficult, and some customers never crack open the book to set it up. Power and flexibility do come with a little complexity, but we’ll always be at your side should you need us.
We know! Rest assured it’s easier for your talent, not harder! 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 the VX, your staff can use the basic stuff immediately because it works just like familiar and comfortable Telos gear. The color, hi-rez LCDs and seamless console integration (to Axia Element and iQ) enhance the user experience. As you read this, systems around the world are screening calls and putting them on the air without drama. Jocks and Talk hosts alike praise the VX! Operators familiar with our longstanding two-column line selection will be right at home.
Yes. We call these “fixed” lines, and you can have as many as you want. They are used for callers who stay on-air while other callers are coming and going.
This is one of the strengths of the VX. Since VoIP calls are four-wire, multiple lines can be conferenced with very high quality. The user interface lets operators assign selectable lines to multiple faders.
Yes – that’s the ultimate. You start with the most flexible console/audio-platform and then add smoothly integrated phones with the IP network powering it all. Sweet! 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.
The VX lets you leverage cheap networking to serve your entire facility. Since you don’t need hardware boxes for each studio, cost is surprisingly reasonable. You’ll use the 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 your Telco service might save you a lot of scratch. We seen stations saving thousands of dollars a month (no kidding) by eliminating POTS lines, with their taxes and fees.
VX is simpler than a multiple-box approach. With fewer components, it’s more reliable. At the time of this writing, no Vx Engine at a radio station has crashed, ever. We’ll probably have a contest to see who has the longest “uptime”. When this was written, it’s 6 months - but that system was installed 6 months ago…
Up to 100! This includes VSet phones, Console Controllers, VX Producer software connections or 3rd-party call management software.
Yes, iQ6 can use multiple VSets along with VX Producer and the iQ Telco Expansion frame. Up to 12 control clients are allowed for each iQ6.
There are thousands of PBXs in use - it would be impossible to test each one! VX uses the Session Initiation Protocol (SIP) standard for all telco connections, so it will work with any PBX that also adheres to the SIP standard, or (using a POTS telco gateway) with any PBX that can "break out" an analog phone line.
Yes, you can use the Telos VX with POTS lines. As with most modern phone systems, POTS lines need to get converted into SIP format. This is done one of several ways: Use an external POTS-to-SIP Gateway. There are several on the market ranging from $200 to $2,000 or more. We suggest using Patton gateways, but other other popular brands such as Audiocodes, Sangoma, Adtran, and Grandstream can work well, too. Our VX configuration experts can help you choose the right model. Use an open-source PBX with POTS adapter cards. This is typically an Asterisk-based phone system, with Digium POTS adapter cards in a PC server chassis. Going this route has the benefit of adding Voice Mail, Legal Disclaimer Announcements, and other sophisticated call-handling options to your Telos VX Studio Talkshow System. Have your business phone system convert POTS to SIP. Modern business phone systems can do this with the right options and programming. Another option is to change the way your phone company delivers these POTS lines to you. POTS, being an analog transport, is prone to audio problems that just don't exist with digital transports. Options from your telco are usually T1, PRI, and SIP. You'll find the caller audio quality to be higher when your calls are delivered in any of these ways. T1 and PRI will still require conversion to SIP before being brought into your Telos VX; T1 and PRI can be converted just as POTS is converted, using any of the first 3 methods above. Eventually, your phone service absolutely will be delivered over SIP, perhaps via a dedicated DSL or other IP transport media. SIP really does simplify the hardware requirements at your studios, so you may want to consider making the switch to SIP now. The best news is that, no matter which telephony transport you want to use, POTS, T1, E1, ISDN BRI, ISDN PRI, or SIP, we're able to walk you through the process. Telos has been through this process with customers hundreds of times, and we're happy to assist you as well. In the end, you'll be delighted with the improved quality of your phone calls, especially when you go to SIP directly.
One might think that lossless, redundancy-reducing methods (such as those used for computer hard-disk compression) would be ideal for audio. Unfortunately, there is not enough redundancy in the audio signal for the significant reduction required by ISDN. Early coding schemes, such as Adaptive Delta Pulse Code Modulation (ADPCM), take advantage of the fact that it takes fewer bits to code the difference, or delta, between successive audio samples compared to using their individual values. Further efficiency is had by adaptively varying the difference comparitor according to the nature of the program material. However, the reduction power of ADPCM is insufficient for transmitting full bandwidth audio over ISDN, as evidenced by 7kHz codecs that use G.722. To develop coding algorithms with sufficient power to achieve the desired reduction, the audio industry has turned to psychoacoustics. Using carefully researched psychoacoustic principles, coding processes have been designed to reflect the way in which human hearing interprets audio information.
It’s a built-in firewall, isolating the VoIP connection from your studio network. We use this same approach in the iPort Livewire-WAN MPEG gateway.
Right. Current Cisco VoIP phones, for example, support the g.722 codec. The VX supports this, as well. However, Steve was probably referring to “AMR Wide Band” also known as G.722.2., sometimes called “HD Audio”. It’s 7khz and doesn’t sound at all like “phone audio” - in fact, it sounds better than regular G.722! AMR-WB is part of the new ITU standard for mobiles, so should grow over time. Meanwhile, some IP-based apps for mobiles are starting to use wideband codecs, such as MPEG-ELD in Apple’s ‘facetime app.
Unlike ISDN, SIP trunks are 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.
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.
If your existing PBX cannot provide a 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. If you do not need to connect VX to your office PBX, however, you might opt for pure SIP Trunking service. We have experience with several SIP Trunking providers, such as EndStream Communications and Bandwidth.com, but you can obtain service from any VoIP provider that will give you a “my own device” plan. Be aware, though, that a simple Vonage account will not work – they limit the number of calls you can take based on the number of trunks you have (usually, in this case, just one). This type service is a home phone replacement — not a true SIP provider.
Since VoIP depends on SIP, and SIP is a defined standard, the only question is if your PBX can provide a trunk 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. An easy way for you to find out whether your PBX will work with VX is to download X-lite, a "softphone" client for the PC. Install and configure it to work with your PBX; if X-lite works with your PBX, VX will too. X-lite can be downloaded for free from http://www.counterpath.com/x-lite-download.html.
Yes, VX works with these by means of “gateways” made by many third-party providers such as Grandstream, Patton, and others. See the “Gateway Products and Suppliers” section of the VX User Manual for some suggestions.
If you have ISDN now and want to keep it, again 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.
You can do that, and it’s not difficult. You only need a POTS gateway device. However, we encourage using digital delivery for the best sound quality.
Without data rate reduction, high-quality audio requires a transmission capacity of about 700kbps for each audio channel. Channels that can handle data rates that high are very expensive and hard to get. More affordable and accessible channels, such as the two 64kbps channels in each ISDN circuit, offer a rate of only about 9% of that of a compact disc. That means you must do some coding to get "12 gallons of water into a one-gallon container." (Note that some refer to coding as "compression" but are generally referring to the same process.)
Done! The VX has it.