Audio conferencing quality issues explained

All digital communications follow a path from the origin of the communication to the destination. A typical digital communication path is made up of four stages:

1. Encoding the audio at the source and creating a data packet

2. Queuing of the data packet

3. Crossing network elements

4. Receiving and processing the data packet

Each stage along this path influences the quality of the audio conference call. Below are some of the most common quality issues experienced in audio conferencing and how they relate to the communication path.

Delays, talk-over and echoing

A user experiencing delays or talk-over in an audio conference could be having a latency problem. Latency is the time that passes between transmitting a data packet and the packet being received. After the data packet has been created and placed into the queue, it begins crossing the elements of a data network. These network elements include the hardware, software, system protocols and the connection medium used to transfer the data.

Each network element will contribute to latency, so the more elements that are involved, the more latency there will be. Latency can be influenced by many factors, such as network capacity (bandwidth), network congestion, quality of service (QoS), number of internet hops and equipment between the two connections.

Talk-over occurs when one participant speaks and a second participant, who has not heard the first participant yet also speaks. The resulting collision of the voices is called talk-over. Another way to look at this is the delay or gap between the person speaking and the other participants hearing the voice.

Echoing in an audio conference can also be caused by latency.

Choppy audio or dropped communication

Conference calls with choppy audio or dropped audio portions are experiencing packet loss. As mentioned above, the audio data is carried over the internet as packets. Good voice quality requires the largest number of packets possible to arrive at the destination. Too many missing or late-arriving packets are immediately noticeable: voices are choppy or cut off altogether, making the conversation hard to understand. The worst case is when so many packets are lost that a participant or participants are dropped from the audio conference.

The solution to packet loss may not be easy to determine. In general, network congestion is the main cause of packet loss and a good starting point. If packet loss is experienced in more than 1% of the audio conference calls, further investigation is recommended.

Robotic or metallic voices

Most users of digital communications have had experience with robotic or metallic-sounding voices. This is known as jitter. It means that there is a time variation between when the voice packets should have arrived and the moment of their actual arrival. Essentially, an audio conference with jitter will have voice packets that don’t arrive at regularly spaced intervals.

Voice packets are sent in an evenly spaced, continuous stream. High-quality audio conferencing relies on this evenly spaced, continuous stream of voice packets. Network congestion, queuing issues and the network elements can introduce variations in the gaps between the packets. The more variations in the gaps between the packets, the higher the amount of jitter in the audio conference call.

Potential solutions

Connect to a high-speed, wired network

High-bandwidth internet linked to a computer via a wired connection offers a better audio conferencing experience than a wireless connection on a less stable network (a Wi-Fi® network, 4G or 5G, for example). Wired connections typically have lower data loss, jitter and latency when compared with wireless and mobile internet connections. The data loss for wireless connections often increases the farther the device is from the signal, like a wireless router or cellular base station, for example.

Check the computer and other device specifications

Choppy or robotic-sounding voices can also be caused by slow devices. Ensure the computer being used for audio and video conference calls has enough RAM, a powerful enough processor, sufficient audio and video cards and a properly configured network card. Cables and cable connections should also be checked to ensure they are not compromising audio quality. The requirements for each device, cable and piece of software used should be checked to ensure all recommendations are being met.

Adjust UC&C platform settings

If one or more participants inject a high noise level into the conference session — such as when calling from a cellphone in a car or by having a loud HVAC system — garbled or watery audio may occur. This is often caused by the UC&C client’s audio processing settings. Adjusting settings like noise suppression, noise reduction, echo cancellation and microphone sensitivity will help solve the garbled audio.

Having audio conference participants mute their microphones one at a time will help determine which participant’s setup is causing the issue. When identified, that participant can adjust UC&C settings to reduce or eliminate the issues.

Investigate the internet connection

If a participant in an audio conference call is hard to hear or experience the issues mentioned in this article, it may be because of internet connection problems. A speed test, a ping test or using a network diagnostic tool can help diagnose the issue or issues. Finding and correcting sources of network congestion, inconsistent bandwidth and other internet reliability issues will improve the quality of audio and video conference calls.

Conferencing platforms like Microsoft® Teams and Zoom, for example, often can analyze a completed conference call and advise if there were issues with internet connectivity. If the issue cannot be found by other means, consult with your internet provider or network administrator.

Avoid high-traffic times

Planning the audio conference call at a different time will help determine if network congestion is an issue. Conference calls scheduled during high-traffic times — such as when there are network backups or regular, larger conference calls are being held — may experience network congestion. Try scheduling your audio conference calls outside of times with high network congestion to see whether there’s an improvement.

Try a test conference call

Conduct a test conference call, asking the participants to join one at a time until the quality issue surfaces. This will help pinpoint whether the issue is being caused by a participant’s system or the one in the room. This will also help determine whether there’s a capacity concern with the network, conferencing setup or both.

Other potential remedies

• Ask participants who are experiencing packet loss to turn off their conferencing video.

• Adjust the settings of the various hardware elements.

• Try adjusting network settings to favor real-time traffic and real-time devices.

• Ask participants to mute their microphones when they are not speaking if echo is an issue.

• Encourage individual participants to use push-to-talk features to help control talk-over.

• Consult troubleshooting articles from the UC&C platform provider.