ADSL noise margins?

[altis]

Anyone know what reasonable figures are?

We're having lots of probs at work with the ADSL connection. Checking out the modem, a D-Link DSL-300T, I see the following figures:

Modem Status
Connection Status Connected
Upstream Rate (Kbps) 448
Downstream Rate (Kbps) 4768
US Margin 26
DS Margin 8
Modulation MMODE
LOS Errors 0
DS Line Attenuation 46
US Line Attenuation 51
Path Mode Interleaved

It strikes me that the downstream margin of 8 - presumably dB - is a bit low but I don't know. Any ideas?

[pedantic]

Information taken from here suggests a SNR of 8db should be ok.

SNR /SNR Margin

The Signal to Noise Ratio on the max product is more flexible than on traditional adsl. On the older fixed line speed products an SNR Margin of less than 6dB suggested that there was a problem with the line.

With Max a target SNR margin is set on the DSLAM and its your router and the DSLAM that agree the best way of meeting this target margin. The target margin range is between 3dB and 15dB and rises in 3dB increments - although it is normal for it to fluctuate either side during the training period. The default is 6dB without interleaving.

Mine is at 10db, and I sync at 8128.

[cybernetic_tiger]

What issues are you experiencing?

If the connection is intermittent are you seeing a loss of sync to the DSLAM (i.e. does the ADSL light on the front go out when connection is lost)?

If the connection is slow/erroring but not loosing sync it may be a good idea to have interleaving turned off as this will cause retransmissions which intern will slow the connection.

A noise margin of 5 and above is stated as being within tolerance for a DSL MAX connection but the attenuation is a little high (but within tolerance).

How long has this been connected and when did the issues start?

[greencreeper]

These are my figures - would suggest yours is a bit low...

Down
SNR (dB): 16.2
Attn(dB): 51.0
Pwr(dBm): 15.6

Up
SNR (dB): 14.0
Attn(dB): 31.5
Pwr(dBm): 5.8

[altis]

Thanks for your input guys.

The system has been up and running at this location (mostly consistantly) for over a year now. I have set up a Asterisk-based VoIP phone network. The first thing they notice is calls breaking up and sometimes being dropped all together. Yesterday, they completely lost everything including all internet connection but this appears to have been due to a fault at Eclipse (ticket 3045):
http://www.eclipse.net.uk/index.cfm?id=closedoutages

Everything seems to work fine for weeks on end and then there'll be a burst of VoIP trouble. Our biggest problem is identifying where the fault lies. The analogue phone line (which is actually used as a fax line and a BT Redcare line) was installed by BT Wholesale but is 'supplied' by another company. The ADSL service on top of this is again through BT Wholesale but supplied by a third company and the VoIP service on top of that is through a fourth. When things start going wrong everyone denies all responsibility!

I've used the ADSL checker here but, when it works, this always reports no problems:
http://www.adslchecker.bt.com/pls/ad...hecker.welcome

The only other thing I can check is the actual ADSL modem itself. This is awkward because, in order to view its status page, I have to pull it out of Bridge mode and this disables my VPN access - I don't know why.

I've just posted the figures above to see if they're sensible or whether this could be associated with the intermittent problems. It looks like they might be.

Ideally, I'd like to find an ADSL / Ethernet modem that will behave just like a cable modem and be visible behind a separate router. Then I can watch these figures at my leisure.

[cybernetic_tiger]

VoIP can be affected by the smallest of issues, possibly additional noise on the line related to weather conditions or exchange VP capacity issues.

You can check the capacity in the exchange by typing the PSTN number into http://usertools.plus.net/exchanges/.

This will show you if the VP is highly contended.

Issues could also be caused by other non-ADSL equipment attached to the line (i.e. Fax and redcare system)

As for the VPN access, if you have a windows XP machine on the LAN side of the router you could use a VPN pass-through to that PC and temporally terminate your VPN there. Or even better set yourself up a free account at http://www.logmein.com and make the PC available for remote access.

[altis]

I can get into the site using Radmin and port-forwarding on the router. However, this only allows me into the server - which is useful - but not as convenient as free access to the whole network.

[dragon]

Originally Posted by cybernetic_tiger

e interleaving turned off as this will cause retransmissions which intern will slow the connection.

Turning interleaving off might not be a very good idea.

I'd only recommend that if you got a good line...
Interleaving does add to latency but thats because it helps to reduce the number of errors on the line and will actually REDUCE retransmissions as its able to reduce the number of packets that are corrupted and therefore need retransmitting.

Although I suppose with it off although you'd probably have a lower sync-rate the lower latency might benefit the voip

http://www.kitz.co.uk/adsl/interleaving.htm

Also Rate adaptive Dsl max should be ok down about 6db downstream SNR margin, if there to many errors the DSLAM should raise the target SNR profile automatically which would give a higher SNR at the expense of sync rate.

Is there anything else going over the ADSL line at the same time as the voip traffic, how many VOIP calls are going on at once, whats the codec being used?

Could it be a case of there isn't enough upstream bandwidth to do everything that they are trying to do at the same time...

Uploading a file for instance could disrupt the VOIP unless your using QOS to prioritize the Voip traffic

[altis]

Generally there's next to no other traffic and I have set up the switch to prioritize those ports with the phones and Asterisk box. The VoIP provider sets a limit of four simultaneous calls but, most of the time, there's just one or maybe two outside calls. We're using the ordinary G.711a (A-law) codec - which sounds excellent when it works!

[dragon]

Originally Posted by altis

Generally there's next to no other traffic and I have set up the switch to prioritize those ports with the phones and Asterisk box. The VoIP provider sets a limit of four simultaneous calls but, most of the time, there's just one or maybe two outside calls. We're using the ordinary G.711a (A-law) codec - which sounds excellent when it works!

I would have thought it should work fine then, not sure what to suggest next