SIP Protocol & Log Flow Analyzer | iN2 Tools

Parse Asterisk, FreeSWITCH, or raw SIP packets to visualize call flows, trace signaling sequences, and diagnose NAT, registration, and codec negotiation issues.

SIP Protocol Sequence & Log Diagnostic Analyzer

Analyze raw Session Initiation Protocol (SIP / RFC 3261) trace logs from Asterisk, FreeSWITCH, Cisco CUCM, and Kamailio. Render interactive call sequence flow diagrams, inspect SDP media payloads, and diagnose 4xx/5xx errors.

SIP Log Input

Paste your Asterisk PJSIP debug logs (pjsip set logger on), FreeSWITCH Sofia siptrace logs, or raw SIP packets.

Click on any message arrow in the diagram to inspect its headers and SDP payloads.

SIP Message Inspector

Select a message in the call flow to see raw headers and media configurations.

Session Initiation Protocol (SIP) Architecture

SIP (RFC 3261) is the signaling protocol standard for initiating, maintaining, and terminating multimedia voice, video, and messaging sessions over IP networks.

Common SIP Response Classes

  • 1xx Informational (100 Trying / 180 Ringing): Session setup progress.
  • 2xx Success (200 OK): Request accepted; call established.
  • 4xx Client Errors (401 Unauthorized / 403 Forbidden / 486 Busy / 488 Not Acceptable): Authentication or codec mismatch.
  • 5xx / 6xx Server & Global Errors (503 Service Unavailable / 603 Decline): Gateway failure or user rejection.

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