You can make both outgoing and incoming calls but you have experienced poor voice call quality. This article discusses the causes of VoIP call quality problems and what you can do to correct them.



It's a common problem of the connection networks or packet switched networks. Because the information (voice packets) is divided into packets, each packet can travel by a different path from the sender to the receiver. When packets arrive at their intended destination in a different order then they were originally sent, the result is a call with poor or scrambled audio.

Solution: Use Jitter Buffers for PBX or IP Phone. (you can refer this topic for Enable Jitter Buffer for Asterisk)


  • A jitter buffer temporarily stores arriving packets in order to minimize delay variations. If packets arrive too late then they are discarded. If a packet was dropped (or simply does not arrive in time) then the receiving device has somehow to “fill in” the gap using a process known as Packet Loss Concealment or PLC. 
  • Packet loss needs to be less than 1% if it is not to have too great an impact on call audio quality. Greater than 3% would certainly be noticeable as a degradation of quality (The default G.729 codec requires packet loss far less than 1 percent to avoid audible errors. Ideally, there should be no packet loss for VoIP)


VoIP delay or latency. In general it is the length of time taken for the quantity of interest to reach its destination. Latency sounds like an echo.

As ITU-T G.114 recommends a maximum of a 150 ms one-way latency. Since this includes the entire voice path, part of which may be on the public Internet, your own network should have transit latencies of considerably less than 150 ms.

There are 3 types of delay commonly found in today’s VoIP networks:Propagation Delay, Handling Delay, Queuing Delay.

Solution: A quality VoIP router can solve many of these issues and will result in business quality Business VoIP Phone Service.

How to determine the Packet loss and Network latency ?

Please run the MTR, which is a powerful network tool enabling administrators to diagnose and isolate networking errors and providing helpful reports of network status to upstream providers. 

It run the best on Linux/GNU Platform, This below is example follow syntax: mtr [option] [destination_host]

  • mrt -rw -c50

The destination host should be the sip domain that your sip account base on.

If your PBX's running on Windows, can run MTR for win, download here. To understand more about MRT please refer at Linode.

Once you get the result, kindly send to

Bandwidth Problem

You will have to ensure that you have sufficient bandwidth for good quality SIP calls. Refer to bandwidth required per SIP calls for more details. If your router provides network statistics, you can easily investigate if your internet capacity is utilized at or near the maximum provided by your Internet Service Provider.

Solution: Upgrade to Business Class High Speed

Hardware Issue

This is one of the most common causes of call quality issues.

  • Router: Many small businesses use their internet connection for both voice and data. This is perfectly fine as long as your router has the ability to prioritize VoIP traffic. Without a router that is configured for packet prioritization, call quality can be impacted by the other users on your network. For example, if during a call, another user on your network downloads a large file, without packet prioritization, your call quality could be degraded. A VoIP router prevents this from happening by giving priority to voice traffic on your network.

  • SIP Device (PBX or IP Phone): Sometimes the SIP hardware goes to crash, or performance overload. In order to confirm issue on hardware, just reboot or use the soft-phone to see how voice quality goes.