Your calling experience can vary from good to poor depending on your network quality. The following metrics can help you test your network quality during calls.
Network quality metrics
| Metric | Description | Why it matters |
| Network Quality | An overall assessment of the suitability of the current network conditions (audio and video). | Good network quality enables clear, uninterrupted calls. Poor quality can result in audio dropouts, delays, distortion, or call failures. |
| Peer | The endpoint involved in the media connection. For 1:1 calls a direct connection with the other user is established. For conference calls the media is shared via a relay or server. | Identifying the peer helps distinguish between:
|
| Connection | Transport method to carry real-time media between call participants or the server. Values explained:
| UDP generally provides lower latency and better real-time performance. Relay and TCP modes increase reliability in restrictive networks. But they usually add delay and reduce quality. |
| Packet Loss | The percentage of transmitted packets that fail to reach the receiving peer. Packet loss reflects network congestion, unstable connections, or interference (especially on mobile networks). | In real-time audio and video:
|
| Ping (Round Trip Time) | The time it takes for a packet to travel from the sender to the peer and back again, measured in milliseconds (ms). It reflects the network latency between two peers. | High ping introduces conversational delay, causing talk-over and unnatural pauses. |
| Jitter | The variation in time between the arrival of consecutive packets. It describes how stable and consistent packet delivery is over time. | High jitter forces larger jitter buffers. This increases latency or causing audio distortion if packets arrive too late. |
Troubleshooting
Corporate networks (office, Enterprise, Virtual Private Network (VPN))
Corporate environments prioritize security and control. This can unintentionally interfere with real-time media traffic.
Common issues
- Blocked or restricted UDP traffic
- Many firewalls allow only TCP/HTTPS, forcing calls to fall back to TCP or relay modes.
- Strict Network Address Translation (NAT) or symmetric NAT
- Prevents direct peer-to-peer connections.
- VPN usage
- Adds latency, increases jitter, and may block UDP entirely.
Symptoms
- Calls connect slowly or fail intermittently
- Audio delay, robotic voice, or dropouts
- Calls consistently use TCP or Relay/TCP
- Good call quality outside the corporate network
Troubleshooting steps
- Contact your system administrator in the corporate networks
- Verify transport mode or connection
- Check whether the call uses UDP, TCP, or Relay.
- Persistent TCP or Relay/TCP usage usually indicates firewall restrictions.
- Test outside the VPN
- Temporarily disconnect the VPN and retry the call.
- Test from an alternate network
- Compare results with a home or mobile hotspot to isolate the issue.
Mobile Networks
Mobile networks are dynamic and optimized for mobility. But are subject to radio conditions and handovers.
Common Issues
- Variable signal strength
- Cell congestion (trains, events, dense urban areas)
- Network handovers (cell-to-cell or between Wi-Fi and mobile)
- Carrier traffic shaping or throttling
Symptoms
- Short audio dropouts
- Temporary one-way audio
- Increased jitter during movement
- Call quality changes mid-call
- Issues that resolve after a few seconds
Troubleshooting Steps
- Check signal quality
- Weak signal increases packet loss and jitter.
- Move to a place with a better signal.
- Switch networks
- Test Wi-Fi versus mobile data.
- Stop background downloads
- Reduce competition for resources.