Real-Time Connection Quality Monitoring
Latency, jitter, packet loss, and resolution are measured every 2 seconds in every call. You'll see connection quality drops instantly.
A bad connection means a bad call — but you can only intervene if you can see it. Bitenta monitors every call's connection in real-time: the browser's RTCPeerConnection.getStats() API is queried every 2 seconds; latency (RTT), jitter, packet loss, resolution, frame rate (FPS), codec, and encryption status are read. A quality score of Good/Medium/Poor/Bad is derived from these metrics and reflected to the representative with a colored indicator in real-time. When quality drops, an alert appears; if the network worsens, automatic reconnection may kick in. With STUN/TURN servers in Turkey and P2P connection, latency stays under 200 ms in most calls.
Measure, rate, alert
Continuous Measurement
Connection is analyzed every 2 seconds with getStats().
Quality Score
Instantly rated as Good/Medium/Poor/Bad.
Instant Alerts
When quality drops, an alert appears on the representative's screen.
What is measured under the hood?
Every call is monitored second by second with the browser’s built-in WebRTC statistics.
Continuous measurement with getStats()
The browser’s RTCPeerConnection.getStats() API is queried every 2 seconds. Latency (RTT), jitter, packet loss, resolution, FPS and codec are read live.
Quality score derivation
A Good / Medium / Weak / Poor score is computed from RTT, jitter and packet-loss thresholds and reflected to the agent instantly with a colored indicator.
Encryption verification
The live statistics confirm that the call is end-to-end encrypted with DTLS-SRTP; the security status is visible at all times.
Alert & recovery on drops
When the score drops, the agent is alerted; if the network worsens, auto reconnect kicks in to keep the call alive.
What you gain by monitoring connection quality
Latency (RTT) & Jitter
Round-trip time and jitter are monitored live in milliseconds.
Packet Loss
Percentage of lost packets is calculated in real-time.
Codec, Resolution & FPS
Used codec, resolution, and frame rate are displayed.
Quality Score
Score of Good/Medium/Poor/Bad derived from RTT/jitter/packet loss.
Instant Alerts
When quality drops, an alert appears on the representative's screen.
Encryption Verification
Live indication that DTLS-SRTP encryption is active.
Every 2 Seconds
Continuous, real-time measurement with getStats().
<200 ms Latency
Low latency with Turkey STUN/TURN + P2P.
Frequently asked questions
Which metrics are monitored?
Latency (RTT), jitter, packet loss percentage, resolution (frameWidth×frameHeight), frame rate (FPS), codec used, and encryption status (DTLS-SRTP). All are displayed live during the call.
How frequently is measurement done?
The browser's RTCPeerConnection.getStats() API is queried every 2 seconds; values and quality score are updated accordingly.
How is the quality score determined?
A level is derived from RTT, jitter, and packet loss thresholds: Good/Medium/Poor/Bad. This score is reflected to the representative with a colored indicator in real-time.
What happens when quality drops?
When the score drops, an alert appears on the representative's screen; if the network continues to worsen, automatic reconnection may kick in. This way, you notice the issue before the call drops.
Can I see if encryption is active?
Yes. The live statistics verify and show that the call is end-to-end encrypted with DTLS-SRTP.
Why is latency low?
The connection is established via P2P and uses STUN/TURN servers in Turkey; this keeps latency under 200 ms in most calls.