Documentation Index

Fetch the complete documentation index at: https://developer.kallglot.com/llms.txt

Use this file to discover all available pages before exploring further.

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SIP Integration

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Prerequisites

• A SIP-capable PBX or SBC
• Network connectivity between your infrastructure and sip.kallglot.com (follow your TLS/TCP profile and firewall rules)
• A Kallglot API key (for programmatic session workflows)
• Developer Portal SIP settings configured (Telephony → SIP / PBX)

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Architecture

1. Programmatic sessions — Your backend calls POST https://api.kallglot.com/v1/sessions with routing.type: "sip" and routes the SIP leg to sip:<session_id>@sip.kallglot.com (or equivalent as documented for your provisioning flow).
2. Inbound-first — Kallglot may authenticate the SIP trunk and associate or create session context consistent with your policy; specifics depend on your routing and numbering plan—validate with support@kallglot.com if you rely on unattended ingress.

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Setup Steps

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1. Configure SIP Credentials

1. Go to Telephony > SIP / PBX and click Configure SIP
2. Enable SIP access using the toggle
3. Note your automatically generated credentials:
   • Username: Your unique SIP username
   • Domain: sip.kallglot.com
4. Configure security:
   • Add your PBX IP addresses to the IP Allowlist (recommended)
   • Click Regenerate Password to create new credentials if needed
5. Copy your password when prompted (shown only once)

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2. Create Sessions via API

async function createSIPSession(callerInfo) {
  const response = await fetch('https://api.kallglot.com/v1/sessions', {
    method: 'POST',
    headers: {
      'Authorization': `Bearer ${process.env.KALLGLOT_API_KEY}`,
      'Content-Type': 'application/json'
    },
    body: JSON.stringify({
      mode: 'bidirectional_translation',
      source_language: 'de',
      target_language: 'en',
      routing: {
        type: 'sip'
      },
      metadata: {
        caller_id: callerInfo.callerId,
        pbx_call_id: callerInfo.pbxCallId
      }
    })
  });

  const session = await response.json();

  // The SIP URI to route the call to:
  const sipUri = `sip:${session.id}@sip.kallglot.com`;

  return { session, sipUri };
}

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3. Configure Your PBX

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Asterisk

; sip.conf
[kallglot]
type=peer
host=sip.kallglot.com
port=5061
transport=tls
username=your_sip_username
secret=your_sip_password
fromdomain=sip.kallglot.com
qualify=yes
disallow=all
allow=opus
allow=ulaw
allow=alaw

; extensions.conf
[translated-calls]
exten => _X.,1,Dial(SIP/kallglot/${SESSION_ID})

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Example: SIP gateway (XML)

<!-- SIP gateway registration block (syntax varies by vendor) -->
<gateway name="kallglot">
  <param name="username" value="your_sip_username"/>
  <param name="password" value="your_sip_password"/>
  <param name="realm" value="sip.kallglot.com"/>
  <param name="proxy" value="sip.kallglot.com"/>
  <param name="register" value="false"/>
  <param name="caller-id-in-from" value="true"/>
</gateway>

<!-- dialplan -->
<extension name="translated_call">
  <condition field="destination_number" expression="^(\d+)$">
    <action application="bridge" data="sofia/gateway/kallglot/${SESSION_ID}@sip.kallglot.com"/>
  </condition>
</extension>

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Cisco CUCM

1. Add a SIP Trunk:
   • Destination Address: sip.kallglot.com
   • Port: 5061
   • Transport Type: TLS
2. Configure SIP Security Profile with digest authentication
3. Create a Route Pattern pointing to the trunk

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4. Handle Call Flow

// When a call comes in to your PBX
app.post('/pbx/incoming', async (req, res) => {
  const { callId, from, to } = req.body;

  // 1. Create Kallglot session
  const { session, sipUri } = await createSIPSession({
    callerId: from,
    pbxCallId: callId
  });

  // 2. Store mapping
  await db.callMappings.insert({
    pbx_call_id: callId,
    kallglot_session_id: session.id
  });

  // 3. Return SIP URI for PBX to route to
  res.json({
    action: 'route',
    destination: sipUri
  });
});

// When call ends
app.post('/pbx/hangup', async (req, res) => {
  const { callId } = req.body;

  // Find and end Kallglot session
  const mapping = await db.callMappings.findOne({ pbx_call_id: callId });
  if (mapping) {
    await endKallglotSession(mapping.kallglot_session_id);
  }

  res.status(200).send('OK');
});

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SIP Configuration Options

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Transport

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Codecs

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DTMF

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Security

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TLS Configuration

1. Go to Telephony > SIP / PBX → Configure SIP → Advanced Settings
2. Enable Require TLS
3. Your PBX must support TLS 1.2 or higher

# Verify TLS connectivity
openssl s_client -connect sip.kallglot.com:5061 -tls1_2

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IP Allowlisting

1. Go to Developer Portal > Telephony > SIP / PBX
2. Click Configure SIP
3. Add your PBX egress IPs to the IP Allowlist (one per line; CIDR supported)

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SRTP for Media

1. Go to Developer Portal > Telephony > SIP / PBX → Configure SIP → Advanced Settings
2. Enable Require SRTP — connections that do not complete SRTP negotiation where required may be rejected (your PBX must negotiate SRTP successfully when this is on)
3. Align crypto suites and RTP/SRTP profiles with your PBX vendor’s documentation; if media fails to establish, collect SIP traces from your environment and work with support@kallglot.com alongside your telephony team.

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Advanced Scenarios

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SIP REFER for Transfers (Coming Soon)

1. End the current Kallglot session
2. Create a new session for the transfer target
3. Route the call to the new session URI

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Multiple Participants (Coming Soon)

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Regional Processing

1. Go to Telephony > SIP / PBX > Configure SIP
2. Under Session Defaults, select your Region:
   • Europe (EU) - GDPR-compliant processing
   • US - Americas
   • Asia-Pacific (AP) - APAC region

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Troubleshooting

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Registration Issues

# Test SIP connectivity
sipsak -s sip:test@sip.kallglot.com -v

# Check firewall allows SIP traffic
nc -zv sip.kallglot.com 5061

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No Audio

1. Check NAT configuration - ensure RTP ports are open
2. Verify SRTP settings match on both sides
3. Check codec negotiation in SIP logs

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Call Quality Issues

1. Monitor jitter and packet loss
2. Check network latency to Kallglot
3. Ensure your region setting matches your network topology

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Debugging

1. Developer Portal > API Logs — recent integration activity (for example webhook delivery attempts). It is not a full SIP signaling trace viewer.
2. Your PBX / SBC logs — SIP dialogs, SDP, RTP/SRTP negotiation.
3. Kallglot support — for authentication or routing issues that need correlation across both environments after you share traces from your side.

# Test SIP connectivity from your PBX
sipsak -s sip:test@sip.kallglot.com -v

# Verify TLS handshake
openssl s_client -connect sip.kallglot.com:5061

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Best Practices

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Security Hardening

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Production Security Checklist

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Credential Security

// Good: Load credentials from environment
const sipConfig = {
  username: process.env.KALLGLOT_SIP_USERNAME,
  password: process.env.KALLGLOT_SIP_PASSWORD,
  domain: 'sip.kallglot.com'
};

// Bad: Never hardcode credentials
// const sipConfig = {
//   username: 'sip_abc123',
//   password: 'hardcoded-password'  // SECURITY RISK
// };

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Network Security

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Contracts, compliance, and data handling

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Policy Configuration

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Session Defaults

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Transport and media policy flags

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Operations checklist

• Invite / setup latency — alert if PSTN callers hear long silence before audio.
• Your PBX counters — failed registrations, 401/403, RTP timeouts vs baseline.
• Concurrent calls vs portal cap — avoid hitting organizational limits mid-day.
• Developer Portal API Logs — correlates webhook/API failures; deeper SIP traces live on your SBC.