SIP Troubleshooting - Poor Voice Quality

When you can make outgoing and incoming calls but the voice call quality is bad, then you experience poor voice quality issue. 

Below are causes of VoIP call quality problems and what you can do to correct them.

Jitter

It is 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)

Note: 

  • 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)

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 sip1.b3networks.com (or the SIP domain in your SIP subscription page)

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 us.


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.

The bandwidth required varies with the codec used. You may check your PBX configuration or hardware provider on which codec you are using for connecting to our SIP server. The values below are estimated per concurrent voice call.

  • G.729: 24Kbps upstream; 24 Kbps downstream
  • GSM: 29 Kbps upstream; 29  Kbps downstream
  • G.711: 80 Kbps upstream; 80 Kbps downstream

The values above are for 1 concurrent voice call. Multiply the values above for each concurrent call you are expecting to estimate your bandwidth required. For example, G.711 codec for 10 concurrent calls, bandwidth required = (80 x 10) = 800 Kbps upstream and downstream.

If bandwidth is insufficient, voice quality will degrade. If you are expecting poor voice quality, please check through the following possible reasons:

Are you sharing the internet?

This is the most common reason for poor voice quality with SIP. If the internet used by the PBX or IP-phones is shared with other computers or machines, the quality drops significantly when these machines are active. Even though you may have very high internet connection speed, a single machine downloading or uploading data on the internet could use up all the bandwidth. This is because most routers distribute bandwidth in a free-for-all manner.

To detect whether your network is experiencing this problem, try turning off all other machines while you make test calls to see if there're improvements to the voice quality.

The best way to resolve this problem is to set QoS on you router. Allocate and reserve bandwidth to you PBX or IP-Phones. Please refer to your router manual about setting QoS. Alternatively, you might need to purchase dedicated internet access for your PBX or IP-Phones, separate from your usual computer networks.

Is your internet connection asymmetric?

You will need equally high bandwidth upstream and downstream for good SIP connections. Your internet connection with your internet provide could be asymmetric. Meaning, your internet connection could be rated, for example 2 Mbps, but you could be getting 2 Mbps downstream and only 128 kbps upstream. Asymmetric internet connection is common in ADSL connections. Please check with your internet provider.

Are you located outside Singapore?

Our SIP servers are located in Singapore. Most internet providers offer overseas connection speed much lower than your rated internet connection speed. For example, a 2 Mbps connection may only connects with 200 Kbps outside your country. Please check with your internet provider. You may also perform a speed test to a Singapore server from www.speedtest.net.

Is your internet speed just enough?

Actual internet connection speed is usually lower than the rated connection speed offered by the internet providers. You should estimate your actual connect speed about 10% lower than the rated speed. 


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.


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