SIP debugging tool

SIP Log Analyzer & Flow Inspector

Paste raw SIP messages or messy Asterisk PJSIP logger output. The tool filters non-SIP log lines, detects SIP messages, groups flows by Call-ID, explains REGISTER and OPTIONS traffic, and highlights SDP codec negotiation for calls.

Client-side validation Raw SIP Asterisk PJSIP

Input

Paste a trace, drop a file, or browse for a log.

Drop log file here or paste directly below.

Results

No SIP messages parsed yet.

Built for real Asterisk PJSIP logs

Asterisk logs often wrap SIP messages in timestamp and logger metadata. This tool removes wrappers, keeps the SIP message boundaries, and ignores unrelated log noise instead of failing the whole parse.

REGISTER and OPTIONS aware

REGISTER and OPTIONS traffic is not a normal call, but it is still a SIP flow. The analyzer summarizes authentication challenges, success responses, failed qualify checks and repeated keepalive transactions.

Clean export for other tools

Use the clean SIP view to copy or download only SIP messages, without syslog prefixes or Asterisk log lines. This makes traces easier to paste into SIP syntax highlighters and diagnostic tools.

How to use it

Analyze a trace

  1. Paste a raw SIP trace, Asterisk pjsip set logger on output, syslog excerpt or journal output.
  2. Use the flow selector to inspect one Call-ID, or keep “All flows” selected for a global overview.
  3. Open a flow to inspect transactions, responses, authentication challenges and negotiated codecs.
  4. Export clean SIP if you need to share the trace or open it in another diagnostic tool.
Privacy note

Browser-only parsing

This page is designed as a static, browser-only tool. SIP logs can contain phone numbers, IP addresses, usernames, authentication headers and customer information. Remove sensitive data before sharing exported traces.