AI-Powered SIP & Freeswitch Analysis

Automatically analyze, debug, and optimize your VoIP configurations with artificial intelligence.

Features

Powerful Tools for VoIP Professionals

Our AI assistant helps you identify and fix issues in your SIP and Freeswitch configurations.

AI Config Analysis

Deep analysis of SIP and Freeswitch configs with syntax and semantic checking. Identifies common pitfalls and misconfigurations.

NAT Traversal Codec Issues ACL Rules

Log File Debugger

Parse and analyze SIP logs to identify call failures, registration issues, and media problems with clear explanations.

408 Timeout 503 Unavailable RTP Timeout

Explanation Engine

Translates complex SIP errors into plain English with actionable recommendations and links to relevant documentation.

Beginner Friendly Expert Mode

SIP Flow Visualizer

Interactive visualization of SIP call flows with color-coded messages and timing diagrams to understand complex scenarios.

INVITE Flow REGISTER Flow

Best Practice Score

Rates your configuration against industry best practices and provides specific optimization suggestions.

Security Performance Reliability

Version Comparison

Compare different versions of your configuration files to track changes and identify potential regressions.

Git Integration Change Tracking

Workflow

How It Works

Simple steps to analyze and optimize your VoIP configurations

Upload your SIP or Freeswitch configuration

Click to upload or drag and drop

XML, CONF files up to 10MB

sip_profiles/internal.xml 
<profile name="internal">
    <settings>
        <param name="debug" value="0"/>
        <param name="sip-trace" value="no"/>
        <param name="context" value="public"/>
        <param name="rfc2833-pt" value="101"/>
        <param name="sip-port" value="5060"/>
        <param name="dialplan" value="XML"/>
        <param name="apply-nat-acl" value="nat.auto"/>
        <!-- Potential issue: Missing stun-server -->
    </settings>
</profile>

Sample Analysis Results

See how SIP Genius helps you identify and fix configuration issues

Critical Issues

Requires immediate attention

NAT Traversal: Missing STUN server configuration

Security: No ACL restricting external access

Warnings

Potential improvements

Codecs: G.711 only, consider adding Opus

Logging: SIP tracing disabled

Best Practices

Configuration strengths

RFC2833: Correct DTMF payload type

Dialplan: Proper XML context

Detailed Issue: NAT Traversal

Your configuration is missing proper NAT traversal settings which can cause one-way audio and registration issues for remote clients.

Recommended Fixes:

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="stun-server" value="stun.freeswitch.org"/>
<param name="aggressive-nat-detection" value="true"/>

Documentation:

Freeswitch NAT Configuration Guide

SIP Flow Visualizer

Interactive visualization of SIP call flows and message sequences

SIP Call Flow Diagram

Upload a SIP log file to visualize the call flow

Pricing Plans

Choose the right plan for your needs

Free

Basic analysis for small deployments

$0 /month

What's included

  • 5 config analyses per month
  • Basic issue detection
  • Community support
  • Log file analysis
  • SIP flow visualization

Professional

For small to medium businesses

$49 /month

What's included

  • 50 config analyses per month
  • Advanced issue detection
  • Email support
  • Log file analysis
  • Basic SIP flow visualization

Enterprise

For large deployments and teams

$199 /month

What's included

  • Unlimited config analyses
  • AI-powered recommendations
  • Priority support
  • Advanced log analysis
  • Interactive SIP flow visualization

Testimonials

Trusted by VoIP Professionals

Sarah Johnson

VoIP Engineer

"SIP Genius saved me hours of debugging by instantly identifying a NAT traversal issue that was causing one-way audio. The clear explanations helped me understand and fix the problem immediately."

Michael Chen

CTO, CloudTelco

"As a CTO overseeing multiple Freeswitch deployments, SIP Genius has become an essential tool in our DevOps pipeline. The best practice scoring helps us maintain consistent, secure configurations across all environments."

David Rodriguez

Senior SIP Developer

"The SIP flow visualizer is incredibly powerful for understanding complex call scenarios. It's helped me explain issues to non-technical stakeholders and train junior engineers on SIP protocol behavior."

Ready to optimize your VoIP configurations? Start your free trial today.

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