VOIP Tech Chat → [General] Line Quality Concern

Hi experts!

I'd like to replace a land line with a VoIP service and have questions about my line quality. Specifically what is acceptable 'jitter' variation?

My research has found that call quality suffers if ping time vary more than 30ms and becomes unusable greater than 100ms.

Testing of my 100/10 connection shows stable 20ms ping times until upload bandwidth is used. Anything greater than 10% - 1MB/s - causes ping times of 40-250ms.

eg:
No load - 20, 20,24,30ms
Load - 48,127,80,230ms

My ISP - Rogers - is unwilling to resolve. I've escalated this issue to 2nd & 3rd - management - level tech support and they continue to state there is no problem they see the can correct. They infer I am creating this problem by "stress testing" their service???

Could I implement a quality service VOIP service these values? If not, how do I push Rogers to correct?

Thanks guys!

What kind of load are you creating to measure your ping times? Do you have any kind of QoS in place that is able to manage your maximum throughput to ensure you don't exceed your available bandwidth?

To ensure you have a good experience with any VoIP service, you want to ensure that you never exceed your available bandwidth even for a brief moment.

If you try sending or receiving more than your pipe can handle, the traffic gets buffered until it has an opportunity to be pushed through. That's what usually causes the jitter.

So when you restrict your traffic to 90-95% of your available throughput, buffering never takes place and this helps keep your phone calls sounding perfect.

That's certainly abnormal -- you shouldn't see a significant increase in ping times until the offered load is greater than upload speed. What kind of modem do you have? Do the signal levels and error statistics look ok? What does a speed test show for upload and download speeds? Does your modem have a router built in and if so, is it enabled? If you are using a separate router, make and model? How are you performing your upstream load test? How is the load source connected (ethernet cable to router, Wi-Fi, other)? Are you pinging from the same device that's doing the upload? If not, how is it connected?

I saw your June 2 speed test with 10.98 Mbps upload result, which seems fine.

By 1MB/s, did you perhaps mean 1 megabyte per second, rather than megabit? If so, that's a large percentage of the available speed and with some normal variations could account for the observed buffering. If that's your case, any router with QoS should protect your VoIP with no trouble.

What kind of modem do you have? Hitron CGN3ACR

Do the signal levels and error statistics look ok? Yes
Rx 2.0- 3.5 mvdB, SNR 40.36
TX 35.7-38.0

What does a speed test show for upload and download speeds?
- 127/11.1 on 100/10 service with intermittent 70/11 hiccups

Does your modem have a router built in and if so, is it enabled?
- enabled built-in router

How are you performing your upstream load test?
- Bitorrent client with upload set to 2 megabits/sec

How is the load source connected (ethernet cable to router, Wi-Fi, other)?
- Ethernet cable or wifi connection makes no difference

Are you pinging from the same device that's doing the upload?
- different computers

I've had two techs visit in the last two weeks and the verify the signal levels are near perfect. Before this post I spent 1.5 hrs on the phone with Rogers who do not acknowledge a problem.

Also there isn't any QoS function enabled on the built-in Rupert.

Does that help provide a clearer picture of my situation? Ask away for anything more.

Hadn't looked at errors before. Here's a look:

No Ranging Response received - T3 time-out
B-INIT-RNG Failure - Retries exceeded
TLV-11 - unrecognized OID
Registration RSP rejected unspecified reason
T6 Timeout and retries exceeded
REG RSP not received

I know nothing about BitTorrent, but suspect that its throttling mechanism may not be doling out the packets evenly, but may be sending in fairly large bursts. (If so, a QoS router would prevent that from causing trouble with VoIP.)

You could capture some traffic with Wireshark on the BT machine to confirm or refute that theory.

Or, (using a wired Ethernet connection) temporarily set the network interface on the BT to 10 Mbps. With no other traffic on the LAN, that should leave you with ~ 1 Mbps and your pings from another machine should go through fine.

Or, if your system handles fragmentation properly, the command
ping -t -l 64000 8.8.8.8
will generate ~ 0.5 Mbps of traffic, up and down. If you run 4 of those (in separate windows), a ping from another machine might be delayed ~50 ms if it came just after a burst of fragments, but you shouldn't see anything approaching the 250 you saw with BT.

Well Stewart I must thank you! Your 3rd option of testing with ping -t -l 64000 8.8.8.8 in 4 different windows and pinging from another computer results in a stable 30ms ping rate multiple times!

Thanks for helping confirm that this ISP connection is stable and can provide a quality VoIP service! Now it's time to configure viop.ms and drop this land line which will result in >$50/mo savings!

It appears that QoS on the CGN3ACR is limited to WMM, so you will need to get a separate router and use the CGN3ACR as a dumb modem. See »www.rogers.com/web/suppo ··· guage=en

In addition to QoS, the new router will need Wi-Fi (or a separate access point), since the modem's Wi-Fi doesn't work when it is in bridge mode.

You may have an old router lying around that will do QoS properly, perhaps by using free third-party firmware (Tomato, DD-WRT, OpenWRT, etc.) However, make sure that it will keep up with your 100 Mbps download speed -- if the router runs out of CPU horsepower on a big download, upstream packets may get delayed and mess up VoIP.