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VoIP Engineering Tools

A free, browser-based toolkit for VoIP engineers. Analyze logs, inspect packet captures, and debug SIP calls — no software installation required.

16
Tools available
100%
Browser-based
0
Data uploaded to cloud
Available tools
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Asterisk Log Analyzer
✓ Ready
Parse and visualize Asterisk / FreePBX log files. See call flows, identify missed calls, trace call-IDs, and filter by context or level.
FreePBX 14–17 Asterisk 18–21 Call flow diagram Multi-theme EN/ES
Launch Tool →
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PCAP VoIP Analyzer
✓ Ready New
Upload a .pcap or .pcapng file and get an instant VoIP health report: auth failures, retransmissions, NAT issues, codec mismatches, and RTP quality metrics.
SIP parsing RTP jitter/loss MOS estimate Issue detection NAT detection
Launch Tool →
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SIP Ladder Diagram
✓ Ready New
Visualize SIP call flows from PCAP files as sequence diagrams — just like Wireshark's Flow Graph, but in your browser. Click any message to inspect the raw SIP.
Sequence diagram Raw SIP inspector Per-call view SVG rendering
Launch Tool →
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SIP Message Inspector
✓ Ready New
Paste any raw SIP message (INVITE, 200 OK, BYE, REGISTER…) and get a color-coded breakdown of every header. Validates structure and highlights potential issues.
Header parser SDP decoder Issue hints 100% client-side
Launch Tool →
Proposed tools — vote for what to build next
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CDR Analyzer
Upload a CSV CDR export from FreePBX or Asterisk. Get charts, peak-hour analysis, top callers/destinations, and duration stats.
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RTP Quality Monitor
Deep dive into RTP streams from PCAP: jitter buffer simulation, MOS per-second graphs, SSRC tracking, and RTCP report parsing.
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SIP Auth Debugger
Debug SIP Digest authentication step by step. Compute expected MD5 hashes, compare with sniffed values, pinpoint wrong credentials or realm mismatches.
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Codec Bandwidth Calculator
Calculate required bandwidth for any VoIP codec (G.711, G.729, G.722, Opus…) with configurable packetization time, overhead, and concurrent calls.
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Dialplan Visualizer
Upload or paste a FreePBX extensions_custom.conf or extensions.conf and get a visual flowchart of the dialplan logic.
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DTMF Decoder
Upload a WAV/MP3 file or paste RFC 2833 RTP payload bytes. Detect and decode DTMF digits with timeline visualization.
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SIP OPTIONS Tester
Send a SIP OPTIONS ping to any SIP server from a backend proxy. Check reachability, measure response time, and view the returned capabilities.
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sngrep Log Parser
Import sngrep capture files or paste sngrep terminal output. Get a clean SIP ladder diagram with one click — no PCAP required.
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E.164 Number Formatter
Normalize phone numbers to E.164 for any country. Batch-convert, detect prefix/format, and validate against number plans. Useful for trunk configs.
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Kamailio Config Linter
Paste a kamailio.cfg and get syntax hints, undefined module warnings, and common security pitfall detection without running Kamailio itself.
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SIP Timing Analyzer
Analyze post-dial delay, ring time, setup time, and response latency from PCAP or log files. Identify where call setup time is being lost.
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FreePBX Backup Diff
Compare two FreePBX backup tarballs and see exactly what changed: extensions, trunks, routes, IVRs — in a clean diff view.