Audio conferencing quality issues explained
Echo, delays, choppy voices, jitter and other issues effecting audio quality
Overview
All digital communications follow a path from the origin of the communication to the destination.
A typical digital communication path will be made up of four stages:
- Encoding the audio at the source and creating a data packet
- Queuing of the data packet
- Crossing network elements
- Receiving and processing the data packet
Each stage along this path influences the quality of the audio conference.
Below are some of the most common quality issues experienced in audio conferencing and how they relate to the communication path.
The Problems
Delays, talk-over and echo
If you are experiencing delays or talk-over in your audio conference, you could be having a problem with latency. The time that passes between transmitting a data packet and the packet being received is latency. Once the data packet is created and placed into the queue it begins crossing the elements of a data network. 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 is. Latency can be influenced by many factors, such as network capacity (bandwidth), network congestion, Quality of Service (QoS), number of internet hops, and any 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 your speaking and the other participants hearing your voice. Echoing in an audio conference can also be caused by latency.
Choppy Audio or dropped communication
Audio conferences 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 to arrive at the destination as possible. 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 or overcome. In general, network congestion is the main cause of packet loss, and a good starting point. If you are experiencing packet loss in more than 1% to 3% of your audio conferencing, further investigation is recommended.
This article from Tom Talks explains the different thresholds for network performance really well.
Robotic or metallic voices
Most users of digital communications have had experience with robotic or metallic sounding voices. If this has happened to you, then you have experienced jitter. This 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, or the different 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.
Potential Solutions
High speed, wired network
A high bandwidth internet connection linked to a computer via a wired connection offers a better audio conferencing experience than a wireless connection on a less stable network (wi-fi, 4G or 5G for example). Wired connections typically have lower data loss, jitter and latency when compared with Wifi 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 voices can also be caused by slow devices. Ensure that the computer being used for audio and video conferences has enough RAM, a powerful enough processor, sufficient audio and video cards, as well as a properly configured network card. Cables and cable connections should also be checked to ensure that they are not compromising audio quality. The requirements for each device, cable and piece of software used should be compared to ensure that all recommendations are being met.
Adjust UC&C client settings
If one or more participants injects a high noise level into the conference, such as a calling from a cellphone in a car or 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 is causing the issue. Once the participant is identified, that participant can adjust the UC&C settings to reduce or eliminate the issues.
Check out this article - Avoiding audio processing conflicts - Teams / Zoom and Nureva HDL
Internet connection
If a participant in an audio conference is hard to hear or experiencing the issues mentioned in this article, they may have internet connection problems. Using tools such as a speed test, a ping test, or 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 can improve the quality of audio and video conferences.
Audio/video conferencing platforms, like Microsoft Teams or Zoom for example, often can analyze a completed conference 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.
- One-way delay maximum: 150ms
- Round trip delay maximum: 300ms
- Jitter maximum (video conference): 20ms
- Jitter maximum (audio conference): 30ms
Conference planning
Planning the conference at a different time will help determine if network congestion is an issue. Conferences scheduled during high traffic times, such as network backups or when regular, larger conferences are held may experience network congestion. Try scheduling your conferences outside of times with high network congestion to see if there is an improvement.
Test calls
Conduct a test call, asking the participants to join one at a time until the quality issue surfaces. This will help pinpoint if the issue is being caused by one of the participant's systems or your own. This will also allow you to understand if there is a capacity concern with your network, conferencing setup, or both.
Other potential remedies
- Ask participants that are experiencing packet loss to turn off their conference video
- Adjusting the settings of the different 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
- Referencing troubleshooting articles from the UC&C platform provider
