Packets Sent
Total count of RTP packets transmitted by the endpoint — the baseline for calculating packet delivery ratios and verifying stream activity.
Packets Sent
| Property | Value |
|---|---|
| Key | packets_sent |
| Unit | Packets |
| Type | Counter |
| Direction | Send |
| RFC | RFC 3550 |
What It Measures
Packets Sent counts the total number of RTP packets transmitted by the endpoint since the start of the media stream. Each RTP packet carries a fixed-duration chunk of audio or video data, so this counter effectively measures how much media has been generated and placed on the wire.
For a G.711 audio stream with 20ms packetization, the endpoint sends 50 packets per second. Over a 60-second call, you would expect approximately 3,000 packets sent. For video at 30fps, the count depends on how many RTP packets each frame requires.
Why It Matters
Packets Sent is the denominator in most delivery calculations:
- Loss ratio — Compare packets sent with Packets Received at the far end to determine how many were lost in transit.
- Stream activity verification — A packets sent counter that stops incrementing means the endpoint has stopped transmitting. This can indicate a hold event, encoder failure, or call teardown.
- Codec and packetization validation — The packet rate should match your codec configuration. If you expect 50 packets per second for G.711 and see 25, something is wrong with the packetization interval.
How CallMeter Measures It
CallMeter counts every RTP packet transmitted by the endpoint. This is a cumulative counter that increments throughout the call, reported per second. The counter is also reported in RTCP Sender Reports as defined in RFC 3550.
Thresholds
This metric does not have fixed thresholds. Expected values depend entirely on the codec and packetization interval:
| Codec | Typical Ptime | Expected Rate |
|---|---|---|
| PCMA / PCMU | 20ms | 50 pps |
| G.722 | 20ms | 50 pps |
| Opus | 20ms | 50 pps |
| H.264 / VP8 | varies | 30-120+ pps |
What Causes Unexpected Values
- Counter stops incrementing — Endpoint stopped transmitting (hold, mute at media level, encoder crash).
- Rate lower than expected — Packetization interval is larger than configured, or the encoder is dropping frames.
- Rate higher than expected — Smaller packetization interval, or the endpoint is sending redundancy (RFC 2198) or FEC packets.
How to Fix It
- Verify stream activity. If the counter plateaus, check if the call is on hold or the media session ended prematurely.
- Check packetization. Compare the actual packet rate against expected rate for your codec and ptime settings.
- Correlate with bytes. Divide Bytes Sent by packets sent to verify packet sizes match codec expectations.
Related Metrics
- Packets Received — The reception counterpart; difference reveals transit loss
- Packets Lost — Packets that were sent but never arrived
- Packets Sent Rate — Per-second transmission rate
- Bytes Sent — Total data volume transmitted