Call quality issues
From VoIP.ms Wiki
There are different factors that could potentially affect the quality of your calls. There are several types of sound issues and these can be related to different causes. We will try to mention here some suggestions, so we can identify which type of issue we are experiencing and what things we need to check to start diagnosing on our own.
Reboot your device
Even if you can browse the internet without any issues and if you think the Internet is working fine, it is possible that something in the network is affecting the calls. The first thing that always needs to be tested is to reboot the ATA device and Router, this way we refresh the connection.
Choose a server
A good recommendation is to send a ping to all the available servers, this way you can verify the latency and pick the best option available for your network. (This is just a slight introduction, please refer to our article Choosing Server for more information.
To rule out if your ATA Device or PBX is the source of the issue, you can do a test with a simple software that can be used for the same purpose (make calls).
How to test using Softphones?
- Create a sub account, this way you do not have to alter the settings on your ATA device for the moment.
- Register the softphone using the sub account credentials and make a call, if the issue is the same, the problem can be in our network, if not, then we can start pointing to your device.
- ZoIPer and X-Lite are recommended by the VoIP.ms staff as they are easy to configure. We also recommend Jitsi, with this softphone you can call the Echo test (4443), put the call on pause and verify the jitter and packet loss values. This can be more reliable then sending a ping because the ping may not be prioritized.
One of the main reasons sound issues may occur is based on the traffic or congestion on the network. First thing to try is check if the issue can be duplicated by making an internal call with the provider, for example using an Echo test application (by dialing 4443) or a voicemail.
Some symptoms that can be present because of the lack of bandwidth available:
- Audio cutting in and out (choppy).
- Voice sounding robotic, like if you were talking under water.
- Audio slowing down or speeding up intermittently during the call.
To test if the bandwidth is affecting our calls:
- Disconnect all the devices from the network
- Disable wireless to make sure no one else is using your internet.
- If your router has QoS, disable it.
- If you were using software to download stuff from Internet (e.g. Torrents) wait a few minutes for this traffic to subside.
After following all these suggestions, use a single device and try to make a call If the audio quality is fine, you are probably dealing with lack of bandwidth, and in this case the use of QoS is recommended ( be certain the set up is well done).
Test with all the codecs g711u, g729 and GSM. Sometimes the issues with the audio can be related with the codecs in use, either because the codec we are using is consuming too much bandwidth for our connection, there is also a chance the device we are using is not supporting this codec very well or it works better with a different one. In any case, this test can also help in the diagnostic.
- Check in your Account or sub account settings, which codec you are allowing, you can test allowing one by one, until you get the best result. If using codecs such as G.711 you may try with a lower bitrate codec such as G729a or GSM (if they are supported by your device/software/system).
Check your ISP
After following these suggestions and you still experience sound issues, you may consider contacting your ISP (Internet provider) just to confirm the issue is not related with them.
Common ISP-related issues include:
- Shared connections. Your ISP advertises "up to" some particular speed, but guarantees absolutely nothing as the minimum. Odds are, those megabits per second are shared with other subscribers of the same ISP so when they go online, your connection slows down unpredictably. Your service works very well... except in peak hours when everyone is online, then suddenly "you're breaking up" or experiencing connection problems.
- Asymmetric connections. You have "up to" a few megabits of what your ISP insists is "blazingly fast" speed for downloads, but they forgot to mention that your upload speed is a tenth that figure or worse. Pick up the 'phone and you hear the other party easily, but they say "you're breaking up" and ask you to constantly repeat things.
- Throttling. Sometimes, an ISP knows they've oversold their bandwidth and, if all of their subscribers go online at once, they will have problems handling large downloads. As a means of damage control, they start playing favorites. Hopefully, they put time-sensitive traffic first (such as voip.ms, which only needs about 80 kilobits per second - and not megabits - for each call) and less time-sensitive downloads last, but there is a risk: some of these schemes allocate bandwidth in a sporadic manner, so that a connection is fast for one second and breaking up the next. That's not good for Internet telephone calls of any kind.
If you have applications which purport to send voice free to other users of the same Internet app (Netmeeting, webcam, Skype...) try an Internet-to-Internet call during the time periods when the problems are at their worst. If your webcam audio breaks up too, the problem might not be VoIP but your ISP. Running Internet "speed test" applications to see if the results are varying widely between attempts may also be very telling.
This isn't a guarantee that your ISP will own up to the issues, let alone fix them, but if your ISP is the problem no voice apps will work.
Tones during calls
Another issue related with the quality during your calls, is when you can hear beep tones during a call, like if someone is pressing a button on the phone or trying to dial. This is usually known as "talk-off" and the device is interpreting the voice as a DTMF digit.
Suggestions to follow:
- Upgrade the firmware in your device, sometimes these bugs are fixed in recent versions.
- Change your DTMF Tx Method to InBand (you have to change this setting in your device and in your account or sub account settings). Test if the DTMF tones are working fine, dial 4747 for this test.
- If Inband doesn't work for you, test with DTMF Process INFO and DTMF Process AVT to No, if the options are available in the device.
- Another alternative you can do: DTMF Tx Method: AVT, DTMF Tx Mode: Strict, DTMF TX Strict Hold Off time: 70.
Echo during calls
We have different factors that can cause Echo during the calls, we will review some suggestions to work with:
- Check the volume on the phone is not too loud, it is possible the phone is causing the issue.
- Make a call dialing 4443 for echo test and see if you can reproduce the same situation with this test.
- Again, check the firmware on the device, usually this can help to reduce the echo if you do not have the latest firmware.
- The default gain on some devices, is typically too high and can cause echo. For instance on Cisco PAP devices, you can adjust the FXS Port Input Gain and FXS Port Output Gain, one at a time, in increments of three. You can test using -1 and -11.
- 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 issue, 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.)
- If you use laptop (integrated mic/speakers), echo can be caused by microphone catching noise from speakers. Try lowering MIC Input sensitivity. Using a headset instead of the microphone and speakers will greatly reduce the amount of noise heard by the other party.
You can hear the other party but they can not hear you, and vice-versa. When a situation like this is present, it is know as "one-way audio" and usually it is related with the NAT. The primary cause for one way audio is the NAT enabled device hiding the topology of the customers network. Many legacy devices do not have a built in ALG (Application Layer Gateway), which changes the headers of the VoIP packets (either SIP or MGCP) to allow the customer to preserve their private network topology and allow them to use VoIP service. One-way audio is caused when one side of the RTP stream is not setup or terminated correctly. RTP is the UDP media stream that carries the audio of a phone call on VoIP. Let's try with the following suggestions:
- From the account or sub account settings, select always NAT=Yes (this is the option recommended by VoIP.ms).
- Try using each codec in a separate way, starting with G711u codec only, from the customer portal > Main menu > Account settings > Advanced tab > allowed codecs.
- Only as a test, place the device in DMZ to test if the issue is related with the NAT. do not leave the device in DMZ after finishing troubleshooting.
- Is your router appropriate for VoIP? If you have a router and a modem, try to bypass the router to verify if the issue gets duplicated.
- If your router includes a SIP ALG and/or SPI Firewall setting please ensure that it is disabled. That setting is common in D-Link and Netgear routers. If this does not help make sure you are using the most recent firmware version for your device.
- You can see what port is being used for audio by looking at the UDP port 5060 traffic. The RTP traffic will typically be in the UDP port range 10000-20000.
- Refer to this link: http://myspeed.visualware.com/indexvoip.php . This link is useful to simulate a VoIP call, it lets you choose location and codec and gives you the jitter and MOS (Mean Opinion Score) which is a measure of voice quality.
- Try to use a Softphones like Zoiper and X-lite to try and duplicate the issue, there are free version.
- Reset your device. In some cases this action resolves the issue.
- If you have an ATA device (Linksys) you can work on the following settings:
Under SIP page.
- RTP Packet Size: 0.020
- G729a Codec Name: G729
- G729b Codec Name: G729
Under the Line page.
- NAT Mapping Enable: Yes
- NAT Keep Alive Enable: Yes
- Preferred Codec: G711u
- Use Pref Codec Only: No
Contact your provider
Could it be the case my quality issue resides on the VoIP provider? Yes, it is possible. Some things we can check and specify to provider when opening the ticket are:
- Are the sound issues present only with incoming calls or only with the outgoing calls, or both? Are the sound issues present while dialing 4443 to reach echo test?
- If issues are present only when calling certain areas or specific countries.(please provide the number(s) when opening the ticket)
- If the issue are happening only with the incoming calls, then please route your DID to echo test, and call it from an external provider (preferably landline) and check if the issues appear when doing this test.
- If you are unable to receive incoming calls on one DID, do the DID's call attempts appear on the Call Detail Records? If not, the DID may be broken at an upstream provider level - especially if other-city DIDs on the same account and same telephone handset are working normally.
- For Canada, if the issue is present with outgoing calls only, test using the different options for Routing (value or premium). If the sound issue happens with one route and not with the other, then you need to contact the provider.
Portions of this article have been taken from "How to Troubleshoot Poor VoIP Audio Quality" by Mango. Used with permission.