VOIP replacing analog

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VOIP replacing analog

Post by NECRAT »

Hello Radio Studio guys.

Question. So our college is in the middle of doing a major renovation that involved removing the radio station, and then eventually replacing it.
However in the midst of the move, the contractor guillotine cut the 25 pair cable going to the old telco hybrids. (We had three lines). And then I heard from IT, that a number of analog phone lines were shut down, including potentially them. So we are now looking at possibly replacing the old analog hybrids with some sort of VOIP system. I believe the school had an integrated Cisco phone system now, it's brand new, but not 100% sure. Are there broadcast grade devices that will act as VOIP adaptors? Anything reliable?

This is an area that I admittedly know very little about, and clearly, thanks to these contractors, get to find out in a hurry.

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Re: VOIP replacing analog

Post by radio_guru »

In a small multi-studio environment, go with the Telos VX Prime system. Sure it's costly. But you get what you pay for with the ability to route lines to studios and not have barge-in's like the old 1A2 or Direct systems. And the audio is really good. If calling in by SIP, it's audio is ISDN quality. We use their internal audio processor for caller audio to level out the nastiest nasty and then run them through a better multiband voice processor so the caller is "always there...loud and clear" without moving the fader. It's right there with the Gentner 612.

When we selected the VX 9 years ago we took a long hard look at what was out there. A lot of newcomers have popped up with their versions. Only Telos and Comrex have systems worthy of broadcast. Especially multi-studio environments.

After that, there is JK audio. But those are simple SIP adapters and need a control surface top manage the lines/call director. Too complicated...

Oh, and the VX has its own caller management/screener system. Basic...yes. But it's better than nothing and my talent loves the fact they can touch screen their calls.

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Re: VOIP replacing analog

Post by TPT »

Having to deal with a college IT department, I would budget about $30 a month for a cellular "home phone."

Especially if you do remote broadcasts, particularly sports broadcasts. Can always patch the phone into a hybrid, if you lose network access. Newer AT&T units have internet access through the cell system, we've use them with Bric links if the cell signal is good enough. Good backup for when IT decide3s to swap out a server on the weekend, shooting the campus network down. After all, no-one is using the U's system on the weekend that U meets Tech in football...

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Re: VOIP replacing analog

Post by radiowave911 »

If they have a full Cisco deployment, have them get you either a Cisco VG202XM or a Cisco VG204XM. Both are analog gateways (ATA) for the Cisco systems, the 202 is 2 lines, the 204 is 4 lines. The lines they present appear as a POTS line. This is if they terminate your numbers on their system.

Otherwise, look for either a system that can terminate SIP trunks like the Telos mentioned earlier or a Comrex system, or go to an ATA (Analaog Telephone Adapter) to terminate your SIP lines and give you what generally appears as a POTS (Plain Old Telephone Service) line to most equipment. If they ask if you need the line set up for faxing, and you use anything like the Tieline or similar digital codec for remote broadcasts, then tell them yes. It changes the way the codecs on the SIP side work - a plain SIP line may or may not work for data traffic such as a fax or a modem.
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Re: VOIP replacing analog

Post by spareparts »

If the studio is staying with an analog broadcast phone system behind the Cisco phone system, the Cisco VG202XM or a Cisco VG204XM would work fine. Ask your telecomm system administrator to make sure they are sending CPC signalling when the call ends - got bit by this one myself.

If you are going to jump to the Telos VX Prime (or other SIP enabled system), you should pay attention to the IP ranges used that is being used by the college and your broadcast phone system as SIP does not work well when using NAT, and the cure is an SBC (Session Border Controller)

On a CUCM prior to V9, The setting that you will be paying attention to is the Max Audio Bit Rate, which usually determines the codec used.

V9+, Audio Codec Preference Lists will allow you to manually set the codec, regardless of what the Max Audio Bit Rate is set to.

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