Voice cutting/line issue

[Mitsara]

Hi all,
Today around 6-7pm i started experiencing issues with my voip/ata. On incoming calls voice is choppy and on outgoing calls other party can't hear me. As i'm typing this and testing the line 12AM EST I noticed that my ATA is not able to register properly all of a sudden (voip.freephoneline.ca, if i change to voip2 i go back to having previous issues). Is their some kind of maintenance going on or reason why this started today?

Thank you.

[freephoneline]

Hi

I also have same problem started yesterday on August 05,2015. Please update if any problem with server


Thanks

[Jos]

I also have same problem
find that last night and on the 2015-08-06 11:00 no solution.
I am in Montreal area

[Mango]

For those of you having audio quality issues, please try a ping test to your VoIP server. From a Windows command prompt, the command is:

ping voip.freephoneline.ca -t

Let the ping run for some time, perhaps 30 minutes, then press CTRL+C to end it. Note the number of packets lost. If this is 1% or greater, this could indicate a routing issue between you and FPL.

[Beebop]

I have the same issue. Started Aug 5.

I can hear calls but there is no outgoing voice.

Thanks.

[Beebop]

For those of you having audio quality issues, please try a ping test to your VoIP server. From a Windows command prompt, the command is:

ping voip.freephoneline.ca -t

Let the ping run for some time, perhaps 30 minutes, then press CTRL+C to end it. Note the number of packets lost. If this is 1% or greater, this could indicate a routing issue between you and FPL.

I tried for a few mins and the loss was zero so 0% but I'll run it for 30 mins as you suggested. Thanks

[Beebop]

I could only run it for 15 mins as I have to use my PC for work (didn't know if that would influence the ping)

Lost was 1 so still 0%

[hanke]

I have been experiencing the same problems with basically audio outbound on incoming calls, since yesterday evening. The audio doesnt pass and other end state its like static chirps although the incoming audio is great. Outbound calls the audio outbound works on occasions and then it doesnt.

I changed ATA's thinking it was bad, but same results occur. I have done the ping test as suggested and my loss is 0%. Changing over to voip2 doesnt change the situation. I notice on Status page there is trouble for some Fongo Customers so hopefully my issue is related to it.

My location is Burlington, Ont with Cogeco Cable.

[Mango]

Thank you for trying the ping tests. It is good to eliminate a local routing issue as a cause of the problem.

[freephoneline]

Hi

I changed my server to voip4.freephoneline.ca:6060 as suggested on website and its work fine so my question to mango is should I continue with this setting?

[hanke]

It appears the problem has been corrected. The Status Page shows phone service is good now. I have done a few test calls and reinstalled my old hardware which I thought was defective and all appears to be ok so far.

Hopefully that the server issue was the problem (and for others as well) so we shall see over the next few hours.

[Beebop]

I can confirm it's working for me too now.

I didn't do anything on my end so whatever the problem was has been fixed. Thanks!

[Jos]

It's working for me too now.

[johnsnow]

I've had a few people say my voice is breaking or or sounds far away and with others it was fine. Ping test shows 0% loss, I'll try making a few more calls to see if the issue comes up with the same numbers that first reported the issue.

[mh1983]

I just had the same experience on a call: my voice cut out but the voice was crystal clear on the incoming side. We're used the same Uniden cordless phones we've used since starting to use FPL.

I recently did the following config:

1. Put my modem/router combo in bridge mode; bridged it to a new Asus RT-N600 router
2. Upgrade to latest firmware on router
3. Disabled SIP ALG on the router
4. Set up traditional QoS as per https://help.close.io/customer/portal/a ... sus-router
5. Double checked HT-701 config (redid recently)

Trying to follow as closely as possible the suggestions/checks to perform around here, but always feel like I'm missing something .Is there anything else I should check?
Is there anything I can test out? The new router adds a 5 ghz WiFI band; could that potentially interfere with the cordless phone?

[Liptonbrisk]

You're experiencing upload jitter. Upload affects outgoing voice.

4. Set up traditional QoS as per https://help.close.io/customer/portal/a ... sus-router

That's not completely correct. It's not even listing UDP 6060, which is what you need for voip4.freephoneline.ca

If your Asus router has WAN/LAN Bandwidth Monitor, visit http://www.asus.com/support/FAQ/1008717/
Assign your ATA the highest priority, and give everything else something appropriately lower.
If you have any other device set to "highest", lower it.

Otherwise, visit http://www.asus.com/support/FAQ/1010951/

Service name can be whatever you want.
Use the MAC address of your ATA; select any for the protocol; and select "highest" for the priority.
Assign all other devices on your local network lower priorities, if need be.

Also read the top half of the post at http://forums.redflagdeals.com/freephon ... #p26986402

[mh1983]

Thanks for the info! I thought with my ability to disable SIP ALG, I could safely give voip.freephoneline.ca a try. But point well-taken on the QoS config not looking quite right.

I updated my traditional QoS as per your recommendation. I wasn't sure what to put in the bandwidth upload and download limit fields, though -- I read 80% of whatever the results are from speedtest.net. Does that sound right?

[Liptonbrisk]

Thanks for the info! I thought with my ability to disable SIP ALG, I could safely give voip.freephoneline.ca a try. But point well-taken on the QoS config not looking quite right.

With Asus routers, having "SIP Passthrough" (the SIP ALG feature in Asus routers) enabled doesn't typically cause problems with Freephoneline.

With other routers, SIP ALG can cause potential one-way audio and registration problems. But SIP ALG is not the source of this jitter issue.

I updated my traditional QoS as per your recommendation. I wasn't sure what to put in the bandwidth upload and download limit fields, though -- I read 80% of whatever the results are from speedtest.net. Does that sound right?

I just want to make a few points before beginning here.

1. If the problem is programs (torrents, other apps, etc.) on your computers, smartphones, gaming consoles, smart thermostats, other internet connected devices, etc., hogging bandwidth, the issue you described is not with download bandwidth. The issue would be upload bandwidth. And that makes sense given that typically ISPs usually offer significantly lower upload speeds than download. You were not experiencing choppy incoming voice.

2. Freephonline's SIP servers are located in Ontario. I'm not sure if they're in Milton, Waterloo (where Fibernetics is located), Burlington, or somewhere else around there, but they're located in Ontario.
http://www.yougetsignal.com/tools/visual-tracert/ suggests voip.freephoneline.ca is located in Milton, Ontario.
Another trace suggests that it's in Burlington.

The further away you are from Freephoneline's SIP servers in Ontario, the greater the chance of something wrong occurring along the path between your ATA and Freephoneline's servers. The greater the distance, typically, the greater (or worse) your pings will be. This affects delay or lag. As pings become greater than 200ms, the potential for crosstalk increases.

Even worse, if you get ping spikes, you will start to encounter bad jitter. Jitter is variance between each successive ping. Jitter is the bane of VoIP. You want low pings and low jitter.

So if you were to ping voip.freephoneline.ca 100 times, and let's say you see 20, 20 19, 18, 20, 22, 24, 30, 12, etc, that would be great. Low pings. Very little variation between each ping.

But if you see 20, 23, 19, 18, 30, 500, 22, 20, 30, 900, that's horrible jitter. You're going to have problems. Audio will begin to break up and sound choppy. And if things get really bad, where you start to experience significant packet loss, the call will drop.

You can also try running http://myspeed.visualware.com/index.php at 8p.m. (especially on a Sunday).
You may be required to install BCS: http://www.visualware.com/bcs/index.html

After visiting the link, choose North America-->Canada-->Ontario-->Toronto. And then select VoIP in the dropdown box on the right.
A MOS score below 4.0 is bad news. It means call quality will not be good.
The advanced (+) tab provides interesting info.

Having a bunch of devices on your LAN hogging bandwidth can cause jitter. That's why QoS can help.
But your ATA can have sufficient bandwidth, and you can still encounter the same problem. See points 3 and 4.

3. Congestion with ISPs usually occurs during prime time around 8 p.m. (can sometimes start earlier around 7:30 p.m.) to 11 p.m. in the evening. Your neighbours in the area will be home during that time, and this is especially true on Sundays. This problem occurs more with cable ISPs, but it can also occur with DSL. With cable, it's referred to as local node congestion. Congestion can increase jitter and pings.

If you're encountering congestion with your ISP, QoS settings are going to do nothing for you. The fault would be with your ISP, and in that case, if you're using cable, switching to another cable ISP usually doesn't resolve the problem. Possibly switching to DSL would (or if you're using DSL, switching to cable might be useful; it really depends on the area someone is living in).

4. The other possibility is that an FPL SIP server could be oversubscribed. If that's the case, then FPL's SIP servers could become overloaded and stop responding as quickly as when they are not being used significantly by customers. However, I'm not noticing any problems lately, and I think if the problem were significant, I would be having issues as well. It wouldn't just be a handful of people complaining. A lot of people would be complaining. If that's the problem, the fault is with FPL. Not necessarily easy to prove though.

5. If the problem is only occurring when using World Credits or making long distance calls, then the issue is likely related to the carrier being used by Freephoneline to the destination.


So, returning to point #1, what you enter for the download limit is not going to address your problem unless incoming audio is choppy. But you said the incoming voice isn't breaking up. Outgoing is. Regardless, yes, you can enter 80% for both, with the understanding that the upload limit is more important for you.

If you believe the problem is your phone, it's simple enough to try a corded one attached to your ATA to see if the problem disappears for good. DECT wireless phones won't have problems, and I think it's unlikely interference is the issue anyway, but it never hurts to test.

[mh1983]

Thought I replied to this, but must've forgotten. Thanks a lot for the helpful info, as always!