Fast SIP/RTP dump diagnostics without sending files to a server
This page mirrors the real workflow: upload, quality summary, filters, call table, SIP/RTP details, export and HTML reports.
Local and fast
Network dumps stay on your PC. The browser reads the files, builds the call list, detects common issues and shows results without slow manual inspection.
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Fast scan
No upload
Dump analysis
This block is used to choose a file, timezone and start the scan. During processing it shows read progress and detected calls.
Drop network dumps in any format here
1 file, 214.1 MB
Files
1
Size
214.1 MB
Calls
311
Dump summary
After analysis, a compact dump-wide summary appears: assessed calls, average MOS/R-factor, good and poor call distribution, RTCP overview and a separate HTML report for the whole dump quality.
Dump summary
Estimate from available RTP/RTCP data, not a full end-to-end MOS.
Assessed
181 / 190
MOS estimate
4.35
R-factor estimate
91.6
RTCP summary
0 / 0
How to read quality metrics
These are quick orientation points for deciding where to look first. Values are estimated from available RTP/RTCP data, so they are quality indicators rather than a lab verdict.
MOS estimate
Roughly a 1 to 4.5 scale. The closer it is to 4.5, the better the expected speech quality.
R-factor estimate
A 0–100 scale. Values around 80 or higher usually look good; lower values mean more degradation.
Loss
The share of RTP packets that did not arrive. Closer to 0% is better; rising loss often sounds like dropouts.
Jitter
How unevenly packets arrive. A little jitter is normal; high jitter makes buffering harder and degrades speech.
Max delta / gaps
The longest pause between packets. Large spikes often mean audible pauses or missing speech fragments.
RTCP
The receiver-side view. Useful when local RTP looks fine but the remote side still reports loss or jitter.
Calls, search and filters
The top of the calls section shows visible calls, critical issues and warnings. Search covers Call-ID, numbers, IPs, codecs, issues, media ports and User-to-User; filters narrow the list quickly.
Calls
Search, filters, issue drill-down and selected call export.
3
shown
1
crit.
4
warn.
All dump issues
Issue chips work as quick filters. You can select multiple issues and keep only calls that need investigation.
All dump issues
Reset · 2 selectedTo see the complete list of issues detected by the service, open the Issues page.
Open the Issues pageColumns
The columns panel lets you hide noise and keep only useful fields: Call-ID, addresses, time, RTP, SIP codes, issues and stream count.
Columns
Call table
The table supports multi-select, sorting, column resizing, drag-and-drop column order and horizontal scrolling for large dumps.
| demo-call-001@192.0.2.10 | +15550101 | +15550201 | sip:+15550201@192.0.2.20:5061 | 192.0.2.10:5060 | 192.0.2.20:5061 | 01/15/2026, 04:21:57 PM | 01/15/2026, 04:22:36 PM | 39s | PCMA, telephone-event | Present | 2 warn. | |
| demo-call-002@198.51.100.10 | +15550102 | +15550202 | sip:+15550202@192.0.2.10:5060 | 198.51.100.10:5060 | 192.0.2.10:5060 | 01/15/2026, 04:22:41 PM | 01/15/2026, 04:23:20 PM | 39s | PCMA | Present | 1 crit. | |
| demo-call-003@203.0.113.10 | +15550103 | +15550203 | sip:+15550203@192.0.2.10:5060 | 203.0.113.10:5060 | 192.0.2.10:5060 | 01/15/2026, 04:23:32 PM | 01/15/2026, 04:24:39 PM | 67s | G729 | No SDP | 2 warn. |
Call details and issues
After selecting a call you see Generated Call-ID, parties, addresses, INVITE, codecs, User-Agent, User-to-User, setup time and issue explanations.
Call details
Generated Call-ID
demo-detail-001@198.51.100.40
From
demo_trunk_a
To
+15550301
Initiator IP
192.0.2.30:5060
Recipient IP
198.51.100.40:5061
INVITE
sip:+15550301@198.51.100.40:5061
Codecs
PCMA, telephone-event
User-Agent
N/A
User-to-User
AB12CD;encoding=hex
Setup
0.32s
RTP
Present
Call issues
Critical packet loss
Stream 192.0.2.30 → 198.51.100.40: 251 packets lost (6.22%).
Critical RTP stream gap
Detected a 1799.94 ms gap between RTP packets.
SDP media addressing
This block shows whether SDP exists, which media IPs and ports were advertised, who declared each port, and how many RTP packets were matched to the call.
SDP media addressing
SDP
Yes
Media IP
198.51.100.40,192.0.2.30
Media ports
21202, 30392
RTP packets
7115
Port ownership
RTP statistics
The RTP table shows streams, payload type, codec, clock rate, SSRC, loss, average and maximum delta, jitter and peak jitter. Threshold violations are highlighted in yellow and red.
RTP statistics
| Source | Destination | SSRC | Payload | Codec | Clock rate | Packets | Loss | Avg. delta | Max. delta | Jitter | Max jitter |
|---|---|---|---|---|---|---|---|---|---|---|---|
| 192.0.2.30:21202 | 198.51.100.40:30392 | 14139745 | 8 | PCMA | 8000 | 3787 | 6.22% | 21.32 ms | 1799.94 ms | 10306.06 ms | 2438469.91 ms |
| 198.51.100.40:30392 | 192.0.2.30:21202 | 61ed6943 | 8 | PCMA | 8000 | 3328 | 9.71% | 22.53 ms | 2475.5 ms | 4.8 ms | 212.72 ms |
Call playback
For a selected call you can open interactive playback: the mixed track is on top and participant tracks are below. Circles on the waveform mark likely audio issues, and the tooltip explains the issue and points to the exact direction.
Mixed track
Letter markers: S, G, J, T
198.51.100.40:21202 → 203.0.113.50:30392
203.0.113.50:30392 → 198.51.100.40:21202
Letter markers
Hovering shows a short non-technical explanation.
Loss or order
Speech may briefly disappear, click or be repeated.
Audio gap
Silence was inserted here, so you may hear a pause or missing words.
Uneven delivery
Speech may sound shaky or stutter because packets arrived unevenly.
Timing jump
Nearby fragments are too far apart, causing a delay or dropout.
SIP signaling and RTP
The flow diagram shows call participants, SIP message direction and RTP streams. It makes responses and actual media flow easier to read.
SIP signaling and RTP
Export and HTML report
Floating actions appear after selecting a call. You can export selected calls to PCAP and generate a detailed HTML report from selected or filtered data.
SIP Analyzer report
311 calls · 1747 critical · 7936 warnings · SIP/RTP snapshot
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