CALL QUALITY / RTCP

Understand why your Asterisk calls sound bad.

Monitor RTP packet loss, jitter and round-trip delay on Asterisk, FreePBX and Issabel. See when quality changed, narrow it down to a trunk or extension, and decide what to check next.

30 days free. No card required. No audio uploads.

A short spike. A useful clue.Investigate this interval

Illustrative example · synthetic measurements, not customer data

Packet loss4.1 %Peak
Jitter48 msPeak
RTCP RTT180 msPeak

Packet loss over 15 minutes

Packet loss in percent. Synthetic samples from 10:00 to 10:15. Peak 4.1%, interval mean 0.8%.4%2%0%
10:0010:0510:1010:15
Reported loss2% diagnostic guide
The mean loss is only 0.8%, but the peak reaches 4.1%. Look at the affected interval and direction before deciding whether the trunk, the local network or an individual phone needs attention.
READ THE SIGNALS

Three measurements. Different clues.

Read the mean, peak and measurement count together. Averages can hide short bursts, while a single peak does not establish a sustained problem.

%

Packet loss

The receiver reports missing RTP packets for an RTCP reporting interval. Loss can coincide with gaps or broken speech.

Investigate at 2%

Compare the same period across trunks and extensions. Check whether the increase is isolated or shared.

ms

Jitter

Jitter describes variation in packet arrival timing. PBXonix converts the RTCP value to milliseconds when the codec clock is known.

Investigate at 30 ms

Check whether bursts line up with busy network periods. Compare wired and Wi-Fi paths where relevant.

ms RTT

Round-trip delay

RTT measures the RTCP round trip. High values can be a clue when callers struggle to take turns. It does not measure one-way voice delay.

Investigate at 300 ms

Compare with the usual RTT for that connection. Check the network route, congestion and the scope of the change.

These are PBXonix diagnostic guide values, not universal voice-quality guarantees. Codec behavior, jitter buffers and the media path matter. Alert thresholds and duration are configured separately in settings.

FROM SYMPTOM TO CHECK

What would you investigate first?

Three worked examples using made-up values. They suggest a direction for investigation; they do not automatically identify a faulty device or provider.

01

Words break up for several users

What you see
Loss peaks at 4.1% · jitter at 48 ms
What it may mean
If several extensions show the same burst on one trunk, investigate a shared media path. If only one extension is affected, start closer to that endpoint.
Next check
Choose the same PBX, time range and media direction. Compare the trunk and extension rows, then check interface errors, link load and QoS configuration.
02

Callers keep talking over each other

What you see
RTT reaches 420 ms · loss stays at 0.2%
What it may mean
Rising RTT is a delay clue, even with little reported loss. These measurements alone cannot locate the delay or prove a carrier fault.
Next check
Compare with a normal period and another trunk. Check whether a WAN route, VPN path or bandwidth demand changed around the same time.
03

There are calls, but the charts are empty

What you see
No RTCP observations · values are unavailable
What it may mean
Missing data is not 0% loss. The collector may be waiting for RTCP, media may bypass Asterisk, or a metric may be unavailable for the negotiated codec.
Next check
Check collector status and freshness first. Confirm that Asterisk exposes RTCP events and that the agent has the required monitoring permissions. Then check the media path.
A REPEATABLE WORKFLOW

Go from a complaint to a focused investigation.

  1. Confirm coverage

    Check the last update and RTCP measurement count. A connected agent can still be waiting for quality observations.

  2. Pick the right interval

    Use the last hour, 24 hours or seven days. Compare the complaint time with both the mean and the peak.

  3. Narrow the scope

    Filter by PBX and media direction. Compare available measurements for known trunks and internal extensions.

  4. Follow the incident

    Review the incident timeline, assign an owner and record the checks performed. Compare the same metrics after the change.

Turn a sustained change into an alert.

Enable quality notifications in settings. Choose thresholds, duration and a minimum number of observations, then receive email or Telegram alerts. Review the problem and recovery in the incident centre; use maintenance mode for planned work.

Explore the incident centre ↗

Technical measurements, private conversations.

The quality module uploads numerical minute aggregates and known technical endpoint identifiers. Customer phone numbers, caller names, audio, packet captures and raw SIP messages are not uploaded. The agent connects to the cloud over outbound HTTPS.

Security & data ↗

Know what the figures represent.

Means are weighted by RTCP observation counts, not by calls or packets. Measurements are grouped, not individual call recordings. PBXonix does not calculate MOS. Quiet charts cannot rule out codec, handset or acoustic problems.

Questions about Asterisk call quality

Why can a registered SIP trunk still have poor audio?

SIP registration describes signaling availability. RTP carries the media, and RTCP reports on that media path. A trunk can remain registered while packet loss, jitter or delay affects a conversation. Check trunk state and call-quality measurements together.

Does this work with FreePBX, Issabel, SIP and PJSIP?

PBXonix collects RTCP reporting events exposed by Asterisk. The collection path has been validated on Asterisk 16 and 20 with chan_sip and PJSIP. FreePBX and Issabel support depends on the underlying Asterisk version, event availability, monitoring permissions and media configuration.

Why is jitter or RTT sometimes missing?

Not every RTCP observation contains every metric. Jitter needs a known codec clock for conversion to milliseconds; RTT also depends on report contents and direction. Direct media or missing RTCP can leave the charts empty. Unavailable values are shown as missing, never as zero.

Are the averages calculated per call?

No. Each metric mean is calculated from the available RTCP observations, weighted by their count. A longer call may produce more observations. Measurement counts are not counts of unique calls, and the loss mean is not a packet-weighted loss percentage for the whole PBX.

Will PBXonix upload customer numbers or record audio?

The call-quality module sends numerical aggregates and known technical PBX endpoint identifiers. It does not upload customer phone numbers, caller names, audio, packet captures or raw SIP. You can read the Security & data page for the broader collection model.

Can I receive alerts for packet loss, jitter or delay?

Yes. Quality alerts can be enabled in settings, with thresholds, duration and minimum observation requirements. Email and Telegram delivery use the configured notification channels. The incident centre helps track ownership, technical comments and recovery.

Connect your PBX. Put call quality in context.

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PBXonix / 01:15

See the signal. Know what to check next.

English interface. Subtitles in your language. Demonstration data.

Explore the demo →
Watch the 75-second tour · Overview
  1. Interactive demo · Fictional data · No installation
  2. Choose a situation. Follow the signal from the overview to the incident.
  3. See the unavailable trunk, open the incident and preview the alert.
  4. Notification preview. Nothing is sent.
  5. Compare packet loss, jitter and delay to narrow the investigation.
  6. Check waiting callers, available operators and the longest wait.
  7. Simulate recovery
  8. Example daily report
  9. 30 days free. No card required.

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