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What If the Voice is Lost When Making VoIP Calls?
What If the Voice is Lost When Making VoIP Calls?
20-Sep-2016 14:12

What If the Voice is Lost When Making VoIP Calls?

When you make calls using VoIP-technologies, you often face a situation when the sound of the voice is lost. Most of these problems are observed in the same direction, that is, at one end of the "wire" the subscriber can be heard, the other end he cannot. Moreover, there can be no sound heard on both sides. In addition, there are the following variations:

  • At the beginning of the conversation (then the voice becomes audible);
  • During a call (usually in a couple of minutes after the conversation begins);
  • When you restart your VoIP equipment.

Why may there be such problems? VoIP telephony is a complex structure consisting of two main mechanisms - signaling and media traffic. The signaling makes it possible to establish a connection between two subscribers. The media traffic transmits the audio data, that is, voice. It works by converting voice into data in the RTP-packages, which are then transmitted from one user to another through the Internet connection. In the process of sending traffic, packages can be lost or delayed if the quality of the Internet connection is poor. In these cases, there are problems with audibility during a telephone conversation. It happens that the data packages do not reach the destination. As a result, the sound is completely absent when the conversation takes place. This may be due to incorrectly configured IP PBX.

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To determine the cause of the loss of voice when you call, you must first diagnose the VoIP system. In general, there are several options:

  • The provider’s remote sip server does not send the traffic. Possibly, the connection is not established at the SIP signaling level.
  • Media traffic is not fed to your server. You should check if the required ports are connected to the NAT device. As an option - your firewall does not let the voice data in. In this case, it is advisable to use SIParator, a special-purpose device which prevents the SIP traffic from blocking by the firewall.
  • Traffic arrives at the subscriber server but does not enter the telephone device itself. As a result, the second caller cannot be heard and it seems that there is no sound. In this situation, the reason may be banal - a faulty speaker or a microphone on the phone. It is also important to correctly set up the IP-phone or softphone - you need to specify the size of the voice package as 20 milliseconds.
  • Incorrect configuration of the system. You may have incorrectly entered the address of the destination for RTP-traffic or incorrectly configured codecs.
  • If you have problems with audibility, you should contact your voip provider to have your system tested. It will give you an opportunity to take a ping test to determine the quality of sound.

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