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 to display information about the data.
Analytics are collected per call leg, not per call:
- A one-on-one call between two internal users is composed of two call legs.
- In a conference, each internal participant generates one call leg.
- Only call legs involving your organization’s users are considered.
- The audio quality is assessed using the MOS (Mean Opinion Score), that applies to the incoming audio stream only, and ranges from 1.0 to 4.5:
- Above 4.0 = Good audio quality
- Below 3.6 = Poor audio quality
When Do Call Legs Appear in the Audio Quality dashboard?
- It may take up to 10 minutes after a call ends for it to be displayed in the Audio Quality dashboard.
- Calls made using physical OXE or OXO phones do not appear in the Audio Quality dashboard.
Why Do Some Calls Show “Unknown" Quality in the Audio Quality dashboard?
- Calls made with an ALE Hub deskphone may initially appear with Unknown quality, taking up to 30 minutes after the end of the call for the quality report to process.
- In some cases, the MOS score cannot be retrieved, such as:
- A network issue occurs while the Rainbow client is sending the MOS data
- A third-party client was used for the call
- On Android devices, specific settings may prevent the Rainbow app from sending audio quality data, resulting in Unknown quality calls. More details can be found in Missing Android notifications
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:
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.
- Select the user whose communication you want to analyze or select All users.
- 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.
- 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. - 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 anduse 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.
-Cancel the filter.
•Use the help icon to get further information.
- 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.