Send Bitrate
Outbound bitrate in bits per second — monitor bandwidth usage against SLA limits and verify codec behavior.
Send Bitrate
| Property | Value |
|---|---|
| Key | bytes_sent_rate_bps |
| Unit | Bits per second (bps) |
| Type | Gauge |
| Direction | Send |
| RFC | RFC 3550 |
What It Measures
Send Bitrate measures the outbound data rate of the RTP stream in bits per second. It tells you exactly how much bandwidth the endpoint is consuming for media transmission at any given moment.
This is the bandwidth number that matters for capacity planning. A G.711 audio stream uses approximately 64 kbps of payload bitrate (87 kbps with IP/UDP/RTP headers). An H.264 video stream might use 500 kbps to 2 Mbps depending on resolution and quality settings. This metric shows the actual bitrate on the wire.
Why It Matters
- Bandwidth budgeting — Know exactly how much bandwidth each endpoint consumes. Multiply by concurrent call count for total network capacity requirements.
- SLA monitoring — Verify that media traffic stays within contracted bandwidth limits. Exceeding limits may trigger throttling from the network provider.
- Codec validation — Each codec has an expected bitrate range. Actual bitrate outside that range indicates misconfiguration or unexpected codec behavior.
- Video quality correlation — For video calls, higher bitrate generally means better quality. A bitrate drop may explain why video quality degraded even without packet loss.
How CallMeter Measures It
CallMeter calculates the send bitrate by measuring the total bytes transmitted during each one-second window and converting to bits per second. This produces a real-time gauge that reflects actual bandwidth consumption.
Thresholds
This metric does not have fixed thresholds. Expected values depend on codec and media type:
| Configuration | Expected Bitrate |
|---|---|
| G.711 (PCMA/PCMU) | ~64-87 kbps |
| G.722 | ~64-87 kbps |
| Opus (voice) | ~24-64 kbps |
| H.264 video (CIF) | ~256-512 kbps |
| H.264 video (720p) | ~1-2 Mbps |
| VP8 video | ~256 kbps - 2 Mbps |
Payload vs Wire Bitrate
Codec specifications list payload bitrate (e.g., G.711 at 64 kbps). The actual wire bitrate is higher because of IP, UDP, and RTP headers. For typical 20ms audio packets, headers add approximately 23 kbps of overhead.
What Causes Unexpected Values
- Bitrate lower than expected — Silence suppression reducing transmission during quiet periods, or a variable-bitrate codec adapting to low-complexity audio.
- Bitrate higher than expected — FEC or redundancy packets, header overhead, or a variable-bitrate codec responding to complex audio/video content.
- Sudden bitrate drop — Bandwidth adaptation (the codec detected congestion and reduced quality), or the stream transitioned to a lower-quality mode.
- Bitrate at zero — Endpoint stopped transmitting.
How to Fix It
- Verify against codec specs. Compare the measured bitrate against the expected range for your codec and configuration.
- Account for overhead. Add approximately 40 bytes per packet for IP/UDP/RTP headers when comparing against payload-only codec specifications.
- Check video settings. For video, verify resolution, frame rate, and target bitrate match your test configuration.
- Monitor for adaptation. Some codecs reduce bitrate in response to detected loss. If bitrate drops without a configuration change, the codec may be adapting to network conditions.
Related Metrics
- Receive Bitrate — Inbound bandwidth for comparison
- Bytes Sent — Cumulative data volume this rate is derived from
- Packets Sent Rate — Packet-level view of the same transmission