Call quality issues - VoIP.ms Wiki

Call quality issues

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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.

Contents

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.

Softphone test

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?

Choppy/Robotic voice

Network traffic

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:

To test if the bandwidth is affecting our calls:

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 codecs

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:

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:

Echo during calls

We have different factors that can cause Echo during the calls, we will review some suggestions to work with:

Note: Input Gain = how you sound to the other party. Output Gain = how the other party sounds to you.

One-Way Audio

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:


Under SIP page.

Under the Line page.

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:


Portions of this article have been taken from "How to Troubleshoot Poor VoIP Audio Quality" by Mango. Used with permission.

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