Problemas de audio - VoIP.ms Wiki

Problemas de audio

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== Elegir servidor ==
== Elegir servidor ==
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Es muy recomendable enviar un ping a cada uno de los servidores, de esta forma podemos checar las latencias y elegir la mejor opción. (Esto es solo una breve introducción, para más información puede checar en este link.[[Choosing Server]] )
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Es muy recomendable enviar un ping a cada uno de los servidores, de esta forma podemos checar las latencias y elegir la mejor opción. (Esto es solo una breve introducción, para más información puede checar en este link.[[http://wiki.voip.ms/article/Elegir_servidor]] )
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'''How to test with a softphone?'''
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'''Cómo probar con un softphone?'''
*'''Create a sub account''', this way you do not have to mess with the settings on your ATA device, for now.
*'''Create a sub account''', this way you do not have to mess with the settings on your ATA device, for now.

Revision as of 18:25, 26 August 2011

Hay varios factores que pueden afectar la calidad de audio de sus llamadas. Hay muchos tipos de problemas relacionados a esto y estos pueden ser debido a diferentes causas. A continuación mencionaremos algunas sugerencias para poder identificar el tipo de problema que esta afectando a su dispositivo y como poder hacer un diagnóstico por su propia cuenta.


Contents

Reiniciar su dispositivo

Aún cuando pueda navegar por Internet en su computadora sin problemas, es posible que algo en su red esté afectando sus llamadas. Lo primero que puede probar es reiniciar su dispositivo ATA y ruteador, de esta forma va a renovar su conexión.


Elegir servidor

Es muy recomendable enviar un ping a cada uno de los servidores, de esta forma podemos checar las latencias y elegir la mejor opción. (Esto es solo una breve introducción, para más información puede checar en este link.[[1]] )


Prueba con Softphone

Para eliminar la posibilidad de que su ATA ó PBX sea la causa de este problema, puede hacer una prueba simple con una aplicación usada para realizar o recibir llamadas, normalmente llamada "Softphone".


Cómo probar con un softphone?

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 is 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:

Now, for 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 for this case the use of QoS is recommended, and make sure the set up is well done.

Test codecs

Test with all the codecs such as 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 or there is a chance also 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, you still experience sound issues? You may consider to contact your ISP (Internet provider) just to confirm the issue is not related with them.

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, is know as "one-way audio", and usually is related with the NAT. 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:

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