SIPSymposium API

Programmatically submit SIP traces for AI-powered diagnostic analysis. Integrate SIP troubleshooting directly into your monitoring pipelines, CI/CD workflows, and voice infrastructure tooling.

v1 Pro & Teams Base URL: https://sipsymposium.com

Authentication

All API requests require a Bearer token in the Authorization header. Generate API keys from your API Keys page. API access is available on Pro and Teams plans.

Request header

Authorization: Bearer sips_a1b2c3d4e5f6a1b2c3d4e5f6a1b2c3d4e5f6

API keys are prefixed with sips_. Keys are shown once at creation and cannot be retrieved again — store them securely.

Rate limits

Plan	Requests per day	Reset
Pro	100 per user	Midnight UTC daily
Teams	100 per user, 300 per team	Midnight UTC daily

API calls share the same daily budget as the web analyzer — one quota per user across all interfaces. Teams plans have an additional team-wide pooled cap so a single member can't monopolize the team's daily quota.

Rate limit headers are returned on every response:

X-RateLimit-Limit: 100
X-RateLimit-Remaining: 87
X-RateLimit-Reset: 1713571200

Error handling

Errors return a JSON object with an error field. HTTP status codes follow standard conventions.

200 OK

Request succeeded

400 Bad Request

Missing or invalid parameters — check the error message

401 Unauthorized

Missing, invalid, or revoked API key

403 Forbidden

Plan does not include API access — upgrade to Pro or Teams

413 Payload Too Large

Trace exceeds 500KB limit

429 Too Many Requests

Rate limit exceeded — check X-RateLimit-Reset for reset time

503 Service Unavailable

Analysis service temporarily unavailable — retry with backoff

POST /v1/analyze

Submit a SIP trace for AI-powered analysis. Returns findings ranked by severity, actionable recommendations, and call quality metrics.

Request body

Field	Type		Description
trace	string	Required	SIP trace as plain text. Paste raw SIP messages from your PBX log, sngrep output, or Wireshark SIP decode. Max 500KB.
context.issue	string	Optional	Hint about the symptom you are investigating. E.g. `"one-way-audio"`, `"registration-failing"`, `"calls-dropping"`
context.endpoints	string[]	Optional	Labels for each SIP endpoint in the trace. E.g. `["SBC", "Carrier", "PBX"]`

Example request

curl -X POST https://sipsymposium.com/v1/analyze \
  -H "Authorization: Bearer sips_your_api_key_here" \
  -H "Content-Type: application/json" \
  -d '{
    "trace": "INVITE sip:bob@biloxi.com SIP/2.0\nVia: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776\n...",
    "context": {
      "issue": "one-way-audio",
      "endpoints": ["Softphone", "Carrier"]
    }
  }'

const response = await fetch('https://sipsymposium.com/v1/analyze', {
  method: 'POST',
  headers: {
    'Authorization': 'Bearer sips_your_api_key_here',
    'Content-Type': 'application/json',
  },
  body: JSON.stringify({
    trace: sipTraceText,
    context: {
      issue: 'registration-failing',
    },
  }),
});

const { findings, severity_counts, recommendations, meta } = await response.json();

console.log(`Found ${findings.length} issues`);
console.log(`Critical: ${severity_counts.critical}, High: ${severity_counts.high}`);
recommendations.forEach(r => console.log(' -', r));

import requests

response = requests.post(
    'https://sipsymposium.com/v1/analyze',
    headers={
        'Authorization': 'Bearer sips_your_api_key_here',
        'Content-Type': 'application/json',
    },
    json={
        'trace': sip_trace_text,
        'context': {
            'issue': 'calls-dropping',
            'endpoints': ['Asterisk', 'Telnyx'],
        },
    }
)

data = response.json()
for finding in data['findings']:
    print(f"[{finding['severity'].upper()}] {finding['title']}")
    print(f"  {finding['recommendation']}")

Response

{
  "findings": [
    {
      "severity": "high",
      "category": "media",
      "title": "RFC 1918 private IP in SDP c= line",
      "detail": "The SDP contains c=IN IP4 192.168.1.100 which is not routable over the internet. The remote endpoint cannot send RTP to this address.",
      "recommendation": "Configure external_media_address in your PJSIP transport or set ext-rtp-ip in FreeSWITCH vars.xml to your public IP."
    },
    {
      "severity": "medium",
      "category": "codec",
      "title": "ptime mismatch between offer and answer",
      "detail": "Offerer set a=ptime:30, answerer set a=ptime:20. Mismatched ptime causes jitter buffer issues.",
      "recommendation": "Set ptime to 20ms on both endpoints to match standard packetization."
    }
  ],
  "severity_counts": {
    "critical": 0,
    "high": 1,
    "medium": 1,
    "low": 0,
    "info": 2
  },
  "recommendations": [
    "Configure external_media_address in your PJSIP transport...",
    "Set ptime to 20ms on both endpoints..."
  ],
  "meta": {
    "analyzed_at": "2026-04-16T18:30:00.000Z",
    "elapsed_ms": 1240,
    "plan": "pro",
    "calls_today": 3,
    "calls_remaining": 97,
    "calls_limit": 100
  }
}

GET /v1/keys

List all API keys for the authenticated user. Returns key metadata — prefix, name, usage — but never the full key value.

curl https://sipsymposium.com/v1/keys \
  -H "Authorization: Bearer sips_your_api_key_here"

POST /v1/keys/generate

Generate a new API key. The full key is returned once only — it cannot be retrieved again. Store it securely immediately.

curl -X POST https://sipsymposium.com/v1/keys/generate \
  -H "Authorization: Bearer sips_existing_key_or_session_token" \
  -H "Content-Type: application/json" \
  -d '{"name": "Production monitoring"}'

Note: You can also generate keys from the API Keys page without needing an existing API key — just sign in and click Generate.

DELETE /v1/keys/revoke

Revoke an API key immediately. Revoked keys cannot be restored — generate a new key if needed.

curl -X DELETE https://sipsymposium.com/v1/keys/revoke \
  -H "Authorization: Bearer sips_your_api_key_here" \
  -H "Content-Type: application/json" \
  -d '{"key_id": "uuid-of-key-to-revoke"}'

Findings schema

Field	Type	Description
severity	string	One of: critical, high, medium, low, info
category	string	Finding category: signaling, media, codec, security, compliance, nat, auth, routing, general
title	string	Short summary of the finding
detail	string	Full explanation with specific values from the trace
recommendation	string	Actionable fix for the finding. May be empty for informational findings.

Severity levels

Severity	Meaning	Examples
Critical	Call will definitely fail or major security issue	No codec match, TLS cert expired, 401 auth loop
High	Call likely to fail or significant quality impact	RFC 1918 in SDP, RTP packet loss >5%, SIP 5xx error
Medium	Potential issue that may cause intermittent failures	ptime mismatch, SRTP not used, missing RTCP
Low	Best practice violation or minor misconfiguration	Non-standard ptime, missing SIP headers, deprecated codec
Info	Informational — no action required	Codec list, call duration, endpoint identification

Code examples

Automated monitoring pipeline

// Auto-analyze calls that end with short duration (quality issue signal)
// Using Asterisk AMI + SIPSymposium API

const AsteriskAmi = require('asterisk-manager');
const fs = require('fs');

const ami = new AsteriskAmi(5038, 'localhost', 'admin', 'secret', true);

ami.on('hangup', async (event) => {
  const duration = parseInt(event.duration || '0');
  const cause = event.cause;

  // Analyze short or failed calls
  if (duration < 10 || cause !== '16') {
    const logPath = `/var/log/asterisk/sip_${event.uniqueid}.log`;

    if (fs.existsSync(logPath)) {
      const trace = fs.readFileSync(logPath, 'utf8');
      const result = await analyzeSIPTrace(trace, cause);

      if (result.severity_counts.critical + result.severity_counts.high > 0) {
        alertOncall(event, result);
      }
    }
  }
});

async function analyzeSIPTrace(trace, issue) {
  const r = await fetch('https://sipsymposium.com/v1/analyze', {
    method: 'POST',
    headers: {
      'Authorization': `Bearer ${process.env.SIPSYMPOSIUM_API_KEY}`,
      'Content-Type': 'application/json',
    },
    body: JSON.stringify({ trace, context: { issue } }),
  });
  return r.json();
}

AI voice agent quality monitoring

// After each AI voice agent call, analyze the SIP trace
// Works with Vapi, Bland.ai, Retell, or custom stacks

async function onCallEnded(callSid, sipTrace) {
  const analysis = await fetch('https://sipsymposium.com/v1/analyze', {
    method: 'POST',
    headers: {
      'Authorization': `Bearer ${process.env.SIPSYMPOSIUM_KEY}`,
      'Content-Type': 'application/json',
    },
    body: JSON.stringify({
      trace: sipTrace,
      context: {
        issue: 'ai-voice-agent-call',
        endpoints: ['AI Platform', 'SIP Carrier'],
      },
    }),
  }).then(r => r.json());

  // Log findings to your monitoring dashboard
  await db.insertCallAnalysis({
    call_sid: callSid,
    severity_counts: analysis.severity_counts,
    findings_count: analysis.findings.length,
    top_issue: analysis.findings[0]?.title,
    analyzed_at: analysis.meta.analyzed_at,
  });

  // Alert on critical media issues that affect STT accuracy
  const mediaIssues = analysis.findings.filter(
    f => f.category === 'media' && ['critical','high'].includes(f.severity)
  );
  if (mediaIssues.length > 0) {
    notifyDevTeam(callSid, mediaIssues);
  }
}

Ready to integrate?

Generate your API key from the API Keys page. Pro and Teams plans include 100 API calls per user per day, shared with the web analyzer.

Get API key View plans