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Audio Quality

Related to

Olivier BRAUN
  • Updated

This feature is also available for members with Analytics role.

The Audio Quality section in the analytics covers all audio communication:

  • Web calls.
  • Phone calls over an OXO or OXE system in Computer mode.
  • Phone calls over a Cloud PBX in Computer mode.
  • Conferences.

These analytics are available:

  • In the Global dashboard.
  • In the dedicated "Audio Quality" tab.

Tips: Select the help icon help.svg to display information about the data.

The following rules apply for all analytics below:

  • Analytics are gathered per call leg and not per call. That means that an answered call between two users of the company will show two legs, while for a conference there are as many legs as participants of the company. You are only shown the legs of your users.
  • The audio quality is only measured for the incoming flow. So only the received audio is measured, and not the emitted flow.
  • The indicator used is the MOS, ranging from 1 to 4.5, which indicates the quality of the audio communication. Above 4 is considered to be a good audio quality while below 3.6 is a bad one.
  • In some situations, the MOS for a call leg could not be retrieved and appears as Unknown (eg. a network cut while the Rainbow client was sending the MOS information after the call; or the use of a third-party client). This also currently applies to communications made with Firefox or with a physical device.
    Audio legs with Unknown audio quality are shown in the list of audio legs in the Audio Quality tab but are not shown in any charts and are not taken into account to compute the percentage of good legs.
  • "No audio" indicates that the user could not hear any audio stream.

Web calls / VoIP Phone calls / Conferences (Global dashboard)

The Audio Quality section is split into three metrics, each presented the same way but focusing on different types of audio communication:

  • Web calls
  • VoIP Phone calls for phone calls in Computer mode, on all systems (OXO, OXE, and Cloud PBX)
  • Conferences

For each metric, three views are available:


  • Aggregated view

    The aggregated view is a pie chart that represents the repartition of the call legs grouped by audio quality: good, medium, bad, and no audio.

    At the bottom is indicated the mean MOS over all legs.


  • Quality vs. volume per day

    This view shows together:

    1- The percentage of good legs as a line.

    2- The number of legs with audio quality good, medium, bad, and no audio.

    This view is useful to analyze the impact of the traffic on the audio quality, for example, to spot possible network congestion leading to bad audio quality on busy days.

  • Mean MOS per day

    This view simply represents the mean MOS per day of the selected period.

Audio Quality Tab

Note: This article shows the Audio Quality tab with the User-level of Analytics. Depending on your settings in Analytics and Privacy, you may not be able to search for a specific user or see users' names.
Calls made with a Rainbow Hub physical phone or with a Rainbow client on Firefox are currently not supported and are displayed here with 'Unknown' quality. Calls made with a physical OXE or OXO phone do not appear here.

  1. Select the user whose communication you want to analyze or select All users.
  2. You may select the period you want, or keep the default 'Last 2 days'. Note that, in opposite to other analytics, here you have access to the audio communication of the current day. Calls may take up to 10 minutes after release before they appear in this dashboard.
  3. You are then presented with four graphs that cover the selected user(s) and period:

    • A pie chart that represents the distribution of the call legs grouped by audio quality: good, medium, bad, and no audio. At the bottom is indicated the mean MOS over all legs.
    • A split of legs per Network type
    • A split of legs per Rainbow clients
    • A per-hour view where each dot represents a 1-hour time slot (for example 2023-12-18 from 10 AM to 11 AM). The size of the dot represents the number of legs in the timeslot while the color represents the ration of Good legs (from red for 0% of good legs to green for timeslots with more than 80% of good legs). Use the sliders below the graph to filter the timeslots based on these two values.
    These graphs may be helpful in identifying global quality issues, for example, calls made on Wi-Fi have lower quality than calls over the LAN, indicating a possible lack of performance of the Wi-Fi network.
  4. You have access to the technical information for all audio legs, helping you to better dig into possible issues:

    • Multiple filters are available: click on "+ Add a filter" to add filter and arrow_dropdown.svguse the menu to select options:
    - All audio qualities: Good, Medium, Bas, No audio or Unknbown.
    - Type: incoming, outgoing, of web or softphone or deskphone calls, conference started or joined.
    - Client: select a Rainbow client (Android, iOS, Chrome, Edge, Bowser, Desktop, ...).
    - Network: LAN, Wifi, Cellular or unknown.
    - Codec: select a codec (OPUS, PVMA, PCMU, G722, ...).
    - Sart time: define a start time in a time range.
    - close_filter.svgCancel the filter.
    • help_blue.svg Use the help icon to get further information.
  5. Here are some explanations about the different information:

    • Type: Web call is used for peer-to-peer communication; Softphone call is used for phone calls on OXO, OXE, or Cloud PBX; Conference is used for collaboration in bubbles.
    • Network: one of LAN, Wifi, Cellular, and Unknown
    • Client: one of Android, iOS; Chrome, Edge, Firefox, Opera, Browser (for other or unidentified browsers); Desktop (conference calls on any browser are marked as Desktop due to current restrictions); Unknown.
    • ICE candidates: Before two peers can communicate using WebRTC, they need to use a technique called ICE (Interactive Connectivity Establishment) to gather and identify candidates able to establish a media path between them. This information may be used for example to identify if the call was made at home or from the company network.
    • Packet Loss: The Packet Loss is the ratio of packets sent by the call party but not received by the user. When packets are lost, the user may experience choppy audio or even complete loss of audio.
    • Jitter: When packets are sent, it is not guaranteed that they will be delivered in the same time gaps that were sent. For example, due to network instability, the packets could be delayed, but then arrive in bursts. The difference - or deviation - from the expected interval is called jitter. The higher the jitter, the lower the audio quality is, for example causing audio distortion.
    • RTT: The Round Trip Time is the time between sending a request and receiving the corresponding response. It is thus a good indicator of the latency of the communication induced by the network and other equipment like proxies and firewalls. A high value, for example in case of network congestion, will result in audio communication that will feel less real-time.
    • Underlined elements can be ordered in ascending or descending order.

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