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How to Troubleshoot Poor VoIP Audio Quality

14K views 9 replies 6 participants last post by  nichtevoice 
#1 ·
VoIP with poor audio quality is frustrating, but the good news is that it is typically solveable. This post outlines some things you can do to troubleshoot poor audio quality. It's a long post but has been broken up into several sections ordered by popularity, so you may read the section that best fits your symptoms. If you can't figure out how to solve your problem, please describe your symptoms, the VoIP equipment you are using, and your router, and someone will try to assist you.

Have you solved a problem before that another member is experiencing? Why not help them out and explain how you did it?


Choppy or Robotic Voice

In a huge portion of cases, poor audio quality is caused by a lack of or poor quality bandwidth on the user side. When testing for this, you should see if you can reproduce these problems by making an internal call such as to your VoIP provider's echo test if they have one, or to voicemail, or anything else that does not traverse the PSTN. Here are some symptoms of audio quality issues caused by lack of or poor quality bandwidth:

  • Audio cutting in and out or "choppy".
  • Audio sounding "robotic" or "under water".
  • Audio slowing down or speeding up.
To test for lack of bandwidth, disconnect all devices from your network and disable wireless, to make sure your neighbours aren't using your wireless signal. If your router has QoS, disable it. If you previously had a great deal of inbound traffic from many sources, such as torrent traffic, wait a few minutes to give this traffic a chance to subside. Now, attempt to make a VoIP call. If audio sounds good, your problem is related to lack of bandwidth and can be solved by setting up QoS. If you already set up QoS, perhaps it is not set up correctly. In either case, feel free to post in this forum and ask for advice.

If you are interested in purchasing a router with good QoS support for VoIP, our members recommend the Netgear WNDR3700 or any router with Tomato firmware such as the Asus RT-N16 (be sure to set Unreplied UDP Timeout to 10 and disable SIP helper). We have seen many D-Link routers that do not work well with VoIP.

To test for poor quality bandwidth, open a command or "DOS" prompt and ping your VoIP provider's equipment. In some cases, such as if your VoIP provider's media gateway is at a different location from their SIP switch, a ping test may not be accurate. However, it is a good place to start from. Note that with VoIP.ms, a popular provider used by many members of this forum, a ping test is accurate for this diagnosis.

To ping your VoIP provider, type at the prompt: ping montreal.voip.ms -t

Replace montreal.voip.ms with the hostname of the equipment of the provider of your choice. Note the time of each ping. If your bandwidth is good, each ping should be within a few ms of the others. One or two high pings is fine, but if none are predictable, you have a problem with "jitter". Note that for this test to be reliable, you should do it over the same type of connection as your VoIP device - for example, if your VoIP device is connected to your router via an ethernet cable, also connect the computer performing the test to your router via an ethernet cable. You may wish to leave the test running for 15 minutes. To end the test, press CTRL+C. Note how many packets were lost. 1% or above is a problem and is known as "packet loss".

If you have poor quality bandwidth that is not caused by your equipment, you must consider your ISP as a source of the problem. Try pinging other servers to see whether the problem exhibits itself in all situations or just to specific servers. If the issue only happens to specific servers, you have a routing issue. It is unlikely that your ISP will solve this in a timely manner unless the issue is widespread. In this case, you should change to a different server operated by your VoIP provider.

Sometimes, you can work around issues caused by mild jitter by increasing the "Network Jitter Level" setting on your VoIP device.


Random Beeps or Tones

If you hear random beeps or tones during a call, as if someone had pressed a button on your phone's keypad, this is known as "talk-off" in which a voice (usually a female voice) is interpreted as a DTMF digit.

The most common solution for talk-off is to change your DTMF Tx Method to InBand. Typically this must be done both on your VoIP device and on your VoIP provider's control panel. When you are done, test in two places: by calling your VoIP provider's voicemail or DTMF test, and also to a random company with an IVR.

If you still hear beeps or tones, you should set DTMF Process INFO and DTMF Process AVT to No if you have these options on your VoIP device. If you require remote access to a physical answering machine connected to your VoIP device, test to be sure this still works after you make these changes.


Echo

Echo may be caused by many things, but there are easy ways to solve the most common causes of echo. Try them in order:

  • Set the volume of your phone to "normal". If the other party's voice is too loud, it can actually be picked up by your handset's microphone and transmitted back to the other party.
  • Plug a phone directly into your VoIP device with a short cable, bypassing your house's wiring.
  • Reduce the FXS Port Output Gain and FXS Port Input Gain (if your VoIP device has such settings,) one at a time, starting with output, in increments of three. Note: Input Gain = how you sound to the other party. Output Gain = how the other party sounds to you.
  • If the above does not solve your problem, and you have a Linksys device, verify that Echo Canc Enable, Echo Canc Adapt Enable, and Echo Supp Enable are set to Yes. (These are default settings.)
  • Navigate to the Regional tab of your Linksys device and enable More Echo Suppression. The side effect of this is that it will be difficult to interrupt the other party while they are speaking.
Actually, there is something else you may do, though it is more expensive: use an IP phone instead of an ATA!!


One-Way Audio

In some cases, you can hear the other party but they cannot hear you at all, or vice-versa. If the audio is absolutely silent in one direction, this is known as "one-way audio". This problen is most often related to NAT. If your VoIP provider has settings for NAT, you should turn them on. You should NOT place your VoIP device in DMZ unless you absolutely know what you are doing. Though this may "solve" the problem, using DMZ can be a security risk and it is not a good solution.

If you regularly experience one-way audio part way through the call, it's possible that your router is simply not appropriate for VoIP. Check your router's log if it has one to see if any events coincide with the time of your issues.


Counterfeit VoIP Equipment

Don't laugh...it's more common than you think, especially if you bought your device on eBay. Enter the MAC address of your device at http://www.coffer.com/mac_find/ to look up its vendor. The vendor should match the supposed manufacturer of your device, such as Cisco or Linksys. Additionally, the MAC address displayed on the Info page of the device's configuration should match the sticker on the outside of the device. If you rub the label on the device with your thumb 5-6 times, the printing should not fade or smudge on a genuine device. Finally, if your device is hot to the touch, it could be counterfeit. (Warm is fine, hot is typically a warning sign.)

You may wish to try testing with a softphone to eliminate your VoIP equipment as part of the problem.


Carrier Issues

In significantly fewer cases, poor audio quality can be caused by an issue with your VoIP provider or their carrier. One way to tell is if you only have audio quality issues with incoming or outgoing calls, but not both.

If your issue can be reproduced with incoming calls, use a POTS (plain old telephone service, i.e. non-VoIP) phone to call your number. If your VoIP provider has an echo test, you should route your number to it; if not, wait for your call to reach voicemail. If your issues persist even when your internet connection and equipment are not involved, your issue is with your VoIP provider or their carrier and should be reported.

If your issue can be reproduced with outgoing calls, and your VoIP provider allows you to configure your routing, try using premium routing. If you're already using premium routing, try switching to a lower-cost route. (If an issue occurs only on the expensive routes, this is a problem and should be reported to your VoIP provider.)

Note that while listening to a "test tone" may be useful in troubleshooting, your problem is solved if you can hear human voice properly. This is because packet loss concealment algorithms are based on human voice patterns and may not always reproduce electronic sounds, even when everything is working correctly.


Still have problems?

With proper management, VoIP audio quality can be equal to or even much better than POTS. If you are experiencing an audio quality issue and can't solve it, describe your symptoms, and tell us what VoIP equipment and what router you are using.

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#3 ·
Counterfeit VoIP Equipment

Don't laugh...it's more common than you think, especially if you bought your device on eBay.
Holy crap! More common than you think is right - turns out I bought one! I picked up a Linksys PAP2 off of eBay a while back and although it works it has an annoying background hum that I've never been able to resolve. It never even occurred to me that the thing could be fake but I looked up the MAC address which begins with 0086F8 and it's not even found. Then I tried your smear test on the label and sure enough it does smear when rubbed. Upon closer inspection there are other telltale signs like the plastic seems a bit too cheap, labels are slightly crooked, etc. I took it apart to check out the circuit board and there is nothing with Cisco or Linksys on it anywhere and a couple of the ICs actually have handwriting on them!

Like I said it never even occurred to me that such things would be counterfeited. Live and learn. Oh well, it certainly explains why the audio quality is crap.
 
#4 ·
That sucks! Sorry to be the bearer of bad news :(

Just in case...have you tested with a non-powered phone? I had a phone once that induced an annoying hum even when connected to a POTS line. I accidentally unplugged its AC adapter one day and the hum disappeared. Of course, so did the Caller ID.

Unfortunately, your assumption that the counterfeit VoIP device is causing the problem is probably accurate, but if you're ever faced with a non-powered phone and a few minutes' spare time you could give it a shot.

m.
 
#5 ·
Yeah, my test phone is actually an old non-powered made-in-Canada Northern Telecom phone. This thing is solid - it weighs about 5 pounds and could probably kill someone if you clocked them with it. Its audio quality is also top notch so I'm sure it's not the issue.
 
#6 ·
We are running a business with VoIP. We have had voice quality issues, but lately *knock on wood* they have been OK. The greatest issue we have is the inability to list ourselves in the white pages. Some VoIP companies are not able to list businesses (in certain parts of Canada), so check it out before you switch over. We did check it out in the beginning and we were told it could be done. That was a statement made in error, and we are still paying for it. The new phone books will be printed soon, so we have to make a decision as to whether we are going to stay with VoIP, leave them all together, or just purchase a landline that would offer us a white page listing. At least we'd be in the book! It's not really saving much money though if you are shelling out capital for a line just to be in the phone book. Our customers are often elderly people so they don't go online as they don't own computers.

Hopefully VoIP will get their act together and offer white pages listings to anyone who wants them.
 
#7 ·
If you're having poor audio quality with business VoIP, the troubleshooting steps are still the same.

Just a word of caution that multiple channels of VoIP can saturate an internet connection very quickly. For example, here in Vancouver, our Shaw Cable internet connection's upload speed is 512 kilobits/second. Since VoIP with the G.711 codec uses about 90 kilobits/second, such a connection could theoretically tolerate five calls. However, you'll see much better performance without saturating your connection. I like to treat a connection as if it runs at 75% or even 66% of its rated speed. 75% of 512 = 384, so practically speaking I should expect to have a maximum of four simultaneous outgoing calls with this connection.

This doesn't leave any room for regular internet traffic, so if I required four simultaneous calls, I would need a second internet connection.

m.
 
#8 ·
What we do to avoid saturation is use the queue. We will only accept 3 simultaneous inbound calls into the store. The next call(s) that come(s) in go to the queue. There aren't enough staff to answer too many calls at once anyhow as there are always customers in the store that need help.
 
#9 ·
Mango -

I'm having sporadic "one-way audio" issues with my setup. I'm using a PAP2T with voip.ms. It seems that in about 1/3 of the inbound calls the inbound caller cannot hear us, but we can hear them. Occasionally nether party can hear each other.

My main router is a Linksys DIR-655. I've got a recent firmware installed and the basic configuration only - I haven't set up any forwarding or anything like that. You mention that the router is usually at fault.

What do you recommend in the way of a gigabit router for VOIP purposes? Any hints on how to configure it properly to avoid this issue?

Thanks.

S
 
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