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 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!!
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.
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.