sundance-communications.com/forum Forums TDM Business Telephone Systems Nortel Telephones & Systems BCM Backup

I am installing a BCM 400 with all IP phones. This is my first BCM and first IP phones install. I have 2 concerns.

I am wondering about having backup phones in case the network goes down. Is there a way to connect analog sets to the BCM 400 that's using all IP phones otherwise?

Another concern is QOS. I am wondering if I will need a QOS switch for the IP phones to have good voice quality, or can they plug straight into the BCM and have good voice quality?

How many IP phones are we talking about? How much bandwidth do you have available?
If the IP phones are sharing a network with PCs, you probably will want QoS. The BCM supports QoS using Diffserv, but you will want layer three switches to achieve QoS using that method.
There are other methods of QoS you can achieve using your network switches and routers, if they support QoS.
If you are using IP phones across a WAN connection, Nortel recommends that you set the default codec at G.729 across the network to prevent transcoding.

Here is a link to a bandwidth calculator:
https://www.asteriskguru.com/tools/bandwidth_calculator.php
The BCM will take ASM 8 Global modules which will give you analog stations

As I tell all my IP telephony clients, your telephones will only be as good as your network.

TRy to always use QOS switches. POE is great too (no power transformers at the desk top).

That's exactly what I needed to know.Thanks guys.

There would be 26 IP phones. I am wondering if these can work on a T1, or if they each need a full 64k to function properly. I guess they wouldn't all be on the phone at the same time.

The other concern was configuring the T1. If customer wants DIDs, I believe we need to reconfigure the T1 for PRI. Currently, the voice T1 has 18 analog lines coming out of it. I intended to use the GATM8 cards for these for lines on the IP phones. Does this sound right?

1st:
The phones will work fine using G.729 codec. That compresses the voice to 8kbps. If you are planning on having a WAN connection come into play with the voice traffic, then you will need to plan on packet overhead. If your WAN is using Frame Relay, figure 8 kbps overhead, PPP will have 6 kbps overhead and ATM averages 38 kbps overhead. With 26 IP phones calculate maybe 6 simutaneous calls.
If your phones will just be used in a LAN environment, you won't need the bandwidth calculator or the overhead calculations. As stated earlier, you will want some sort of QoS if the phones are sharing a network. I would suggest using weighted fair queuing.

2nd:
Unless you have the BCM at a remote location from the phones, the T1 doesn't come into play, as far as the phones are concerned. DIDs work fine across a T1.

A PRI will pass CID info where a T1 will not.

The GATM8 is for analog stations. You need to purchase keycodes to activate your IP phones on the system.

If you are planning on keeping the current T1 infrastructure in place, You will need an ATM in the BCM to bring the analog lines into the BCM.

I don't need WAN, as far as I can tell. But, for when I do-
1.ATM as in "Asynchronous Transfer Mode"?
2.Where would I configure the G.729 codec?
3. Vendor says the 16 analog lines will work with the BCM using the GATM8 cards for the lines and purchasing IP Client Licences. Are IPCL same as the keycodes you mentioned?

I am planning on using the voice T1 which brings in the analog lines for the IP phones. I should have at least 10 channels (I think), as 14 channels are bringing in the bundled lines.
(provider says 10 channels available for data, vendor "tech support" says 23 should be available after lines go into system).

I'm still a little shaky with this T1 config and how many channels available , as I'm getting conflicting info from service provider and vendor. It's always tough installing things for the first time!

Yes, ATM is Asynchronous Transfer Mode.

Off the top of my head, in the versions 3.7 and below, it is in Services, IP Telephony, and is under one of the tabs there, maybe general something-or-other. You can change the default codec using the configuration tab from G.711 if you need to. In the 4.0, I think it's under IP Subsystems, under the configuration header.

If your LAN Network seems to have fast switching and you're using QoS, leave it at G.711 and see what happens.

I misread your last post and thought you had typed ASM instead of GATM.

The keycodes and IPCL are the same.

A T1 has 24 channels available sharing 1.544 mbps up and down. Normally the telco will fractionalize the T1 12 and 12, but it sounds like they are giving you 10 channels to use, maybe they are keeping a couple for other things.

As Telemarv stated, try and get a POE switch for your phones to make it easier on yourself. Also you will want to set the phones either in their own network or in their own VLAN.

Sounds like you have an integrated T1, (not a fractional) with 14 voice channels provided as analog dial tone and 10 channels available for data. Your vendor tech support person seems to assume you will have a full voice PRI.