Implementing Trunk Diagnostic Trace Capture and SIP Ladder Diagram Generation Tools

How-To-Articles
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Implementing Trunk Diagnostic Trace Capture and SIP Ladder Diagram Generation Tools

What This Guide Covers

This guide details the architectural implementation of a real-time SIP trace capture and visualization engine for Genesys Cloud CX. You will build a system that intercepts SIP signaling via the Genesys Cloud Platform APIs, parses raw SDP bodies, and renders interactive ladder diagrams to diagnose media path failures, codec mismatches, and SIP 4xx/5xx errors. The end result is a debugging interface that replaces manual log parsing with visual, chronological call flow analysis.

Prerequisites, Roles & Licensing

  • Licensing: Genesys Cloud CX 2 or higher. Genesys Cloud Speech Analytics is not required but recommended for correlating audio quality with signaling events.
  • Permissions:
    • Telephony > Trunk > View
    • Telephony > Trunk > Edit (Required to enable SIP trace logging on specific trunks)
    • Admin > API Key > Create (Required to generate the OAuth token for the integration)
    • Reporting > Report > View (Required to access historical call logs for replay)
  • OAuth Scopes:
    • telephony:trunks:view
    • telephony:trunks:edit
    • telephony:logs:view
    • telephony:logs:download
  • External Dependencies:
    • A middleware application (Node.js, Python, or Java) capable of handling WebSocket connections and HTTP polling.
    • A frontend framework (React, Vue, or Angular) for rendering the ladder diagram.

The Implementation Deep-Dive

1. Enabling SIP Trace Logging on Genesys Cloud Trunks

The first step is configuring the Genesys Cloud trunk to emit detailed SIP signaling logs. By default, Genesys Cloud retains summary call data. To diagnose protocol-level issues, you must enable the SIP Trace feature. This feature captures the full SIP message body, including SDP offers and answers, and stores them in the Genesys Cloud log repository.

Navigate to Telephony > Trunks in the Genesys Cloud admin console. Select the target trunk (e.g., a SIP Trunk Provider or a Genesys Cloud Direct Trunk). In the Advanced Settings tab, locate the SIP Trace section. Toggle Enable SIP Trace to On.

The Trap: Unlimited Trace Retention Costs
Enabling SIP Trace on a high-volume trunk (e.g., >1,000 concurrent calls) generates massive amounts of data. Each SIP dialog can generate 10-20 messages (INVITE, 100 Trying, 180 Ringing, 200 OK, ACK, BYE, etc.). If you leave this enabled permanently, you will incur significant storage costs and potentially hit API rate limits when retrieving logs.

Architectural Decision:
Do not enable SIP Trace globally or permanently. Implement a “Just-in-Time” tracing strategy. Use the Genesys Cloud API to enable tracing only for specific diagnostic windows or for trunks suspected of issues. For production environments, configure the trace retention period to the minimum necessary (e.g., 24 hours) to satisfy debugging requirements without bloating storage.

To enable tracing programmatically for a specific trunk, use the following API call:

PUT /api/v2/telephony/providers/edge/trunks/{trunkId}
Authorization: Bearer {access_token}
Content-Type: application/json

{
  "name": "Production SIP Trunk",
  "enabled": true,
  "sipTraceEnabled": true,
  "sipTraceRetentionDays": 1
}

Key Configuration Field: sipTraceEnabled

  • Value: true
  • Impact: Genesys Cloud begins capturing full SIP message payloads for all calls routed through this trunk.

2. Retrieving SIP Trace Logs via the Genesys Cloud API

Once tracing is enabled, you must retrieve the logs. Genesys Cloud provides two primary methods for accessing SIP trace data: the Telephony Logs API and the Call Logs API. For real-time debugging, the Telephony Logs API is superior because it provides granular, message-by-message data rather than aggregated call summaries.

Querying the Telephony Logs API

The endpoint /api/v2/telephony/logs allows you to query logs based on various filters, including trunkId, startTime, and endTime. To capture a specific call, you must first identify the callId or conversationId from the Call Detail Record (CDR).

Step 2.1: Identify the Call ID
Use the Call Logs API to find the specific call you want to trace.

GET /api/v2/analytics/conversations/details/summary?query=filter:callId eq {callId}
Authorization: Bearer {access_token}

Step 2.2: Retrieve SIP Messages
Once you have the callId, query the Telephony Logs API for SIP messages associated with that call.

GET /api/v2/telephony/logs?filter=callId eq {callId}&sort=timestamp asc&pageSize=100
Authorization: Bearer {access_token}

Response Payload Example:

{
  "pageSize": 100,
  "pageNumber": 1,
  "total": 12,
  "items": [
    {
      "id": "log-12345",
      "callId": "call-67890",
      "timestamp": "2023-10-27T10:00:00.000Z",
      "direction": "outbound",
      "sipMethod": "INVITE",
      "sipStatus": null,
      "from": "sip:user@genesyscloud.com",
      "to": "sip:destination@provider.com",
      "via": "SIP/2.0/TLS edge.genesyscloud.com",
      "body": "v=0\r\no=- 12345 1 IN IP4 192.168.1.1\r\ns=-\r\nc=IN IP4 192.168.1.1\r\nt=0 0\r\nm=audio 4000 RTP/AVP 0 8 9 18 101\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:18 G729/8000\r\na=rtpmap:101 telephone-event/8000\r\n"
    },
    {
      "id": "log-12346",
      "callId": "call-67890",
      "timestamp": "2023-10-27T10:00:01.000Z",
      "direction": "inbound",
      "sipMethod": null,
      "sipStatus": "100 Trying",
      "from": "sip:destination@provider.com",
      "to": "sip:user@genesyscloud.com",
      "via": "SIP/2.0/TLS provider.com",
      "body": ""
    }
  ]
}

The Trap: Ignoring SDP Body Parsing
Many developers stop at the SIP status codes. However, most media failures (one-way audio, no audio) are caused by SDP mismatches, not SIP errors. The body field contains the SDP offer/answer. If the provider rejects a codec in the answer that was offered in the INVITE, the call will connect but have no media. You must parse the SDP body to extract the m= (media) and a=rtpmap= (codec) lines.

Architectural Decision:
Implement a local SDP parser in your middleware. Do not rely on Genesys Cloud to pre-parse SDP data. This allows you to compare the codec lists in the INVITE vs. the 200 OK to detect mismatches immediately.

3. Building the SIP Ladder Diagram Renderer

A ladder diagram is a visual representation of the SIP dialog, showing the sequence of messages between two endpoints (User Agent Client and User Agent Server). It is the industry standard for debugging SIP issues.

Data Structure for the Ladder Diagram

Transform the API response into a structured format suitable for rendering. Each message should include:

  • Timestamp: For calculating latency.
  • Direction: Inbound or Outbound.
  • Message Type: INVITE, 200 OK, BYE, etc.
  • Status Code: For responses.
  • Codec List: Extracted from SDP.
  • Error Details: If applicable (e.g., 488 Not Acceptable Here).

Node.js Example: Processing Logs for Visualization

const processSipLogs = (logs) => {
  const ladderData = logs.items.map(log => {
    let codecList = [];
    if (log.body) {
      // Simple regex to extract codecs from SDP body
      const codecMatches = log.body.match(/a=rtpmap:(\d+) (\w+)\/(\d+)/g);
      if (codecMatches) {
        codecList = codecMatches.map(match => {
          const parts = match.split(' ');
          return { payload: parts[1], name: parts[2], clockRate: parts[3] };
        });
      }
    }

    return {
      id: log.id,
      timestamp: new Date(log.timestamp),
      direction: log.direction,
      method: log.sipMethod || log.sipStatus,
      from: log.from,
      to: log.to,
      codecs: codecList,
      latency: calculateLatency(log.timestamp) // Implement latency calculation based on previous message
    };
  });

  // Sort by timestamp to ensure correct order
  return ladderData.sort((a, b) => a.timestamp - b.timestamp);
};

Rendering the Ladder Diagram

Use a frontend library like React Flow or D3.js to render the diagram. The diagram should consist of two vertical lines representing the Genesys Cloud edge and the external provider. Messages are drawn as horizontal arrows between these lines.

Visual Elements:

  • Green Arrows: Successful messages (e.g., 200 OK).
  • Red Arrows: Error messages (e.g., 4xx, 5xx).
  • Yellow Arrows: Provisional responses (e.g., 100 Trying, 180 Ringing).
  • Codec Mismatch Highlight: If the codec list in the answer does not overlap with the offer, highlight the relevant messages in orange.

The Trap: Ignoring Timestamp Precision
SIP dialogs can happen in milliseconds. If you render messages based solely on array index, you may misrepresent the timing. Always use the timestamp field to calculate the exact duration between messages. A 100ms delay between INVITE and 100 Trying is normal. A 5000ms delay indicates a network issue or provider congestion.

Architectural Decision:
Implement a “Zoom” feature in the ladder diagram. Allow users to zoom in on specific segments of the call (e.g., the initial setup phase vs. the teardown phase). This prevents clutter when dealing with long-duration calls with many SIP re-INVITEs (e.g., for call transfers or conference additions).

4. Implementing Real-Time Trace Capture via WebSockets

For live debugging, polling the API is insufficient. You must use Genesys Cloud’s WebSocket capabilities to receive real-time events. However, Genesys Cloud does not expose raw SIP traces via WebSocket directly. Instead, you must combine Real-Time Call Events with On-Demand Trace Retrieval.

Step 4.1: Subscribe to Real-Time Call Events

Use the /api/v2/analytics/events/realtime WebSocket endpoint to monitor call state changes.

const ws = new WebSocket('wss://api.mypurecloud.com/api/v2/analytics/events/realtime');
ws.onopen = () => {
  const message = {
    "subscribed": ["call-state"],
    "filter": {
      "query": "filter:trunkId eq {trunkId}"
    }
  };
  ws.send(JSON.stringify(message));
};

Step 4.2: Trigger Trace Retrieval on State Change

When a call state changes (e.g., call-state: answered), trigger an API call to fetch the latest SIP logs for that callId.

ws.onmessage = (event) => {
  const data = JSON.parse(event.data);
  if (data.type === "call-state") {
    const callId = data.callId;
    const state = data.state;
    
    if (state === "answered" || state === "failed") {
      fetchSipLogs(callId); // Function defined in Step 2
    }
  }
};

The Trap: Rate Limiting on Real-Time Queries
If you have high call volume, triggering an API call for every state change will hit rate limits. Implement a debouncing mechanism. Only fetch logs when the call ends or when a user explicitly requests a trace for a specific call.

Architectural Decision:
Cache the SIP logs locally in your middleware application. When a user requests a trace for a call that is still in progress, serve the cached logs and refresh them every 5 seconds. This reduces API calls and provides a smoother user experience.

5. Advanced Diagnostics: Codec Mismatch Detection

One of the most common SIP issues is codec mismatch. The Genesys Cloud edge may offer a set of codecs (e.g., G.711, G.729, Opus), but the provider may only accept a subset. If the provider rejects all offered codecs, the call will fail with a 488 Not Acceptable Here. If the provider accepts a low-quality codec (e.g., G.729) while the customer expects HD voice (Opus), the call will connect but have poor quality.

Implementing Codec Comparison Logic

In your middleware, compare the codecs array from the INVITE (offer) and the 200 OK (answer).

const detectCodecMismatch = (inviteCodecs, answerCodecs) => {
  const offerPayloads = inviteCodecs.map(c => c.payload);
  const answerPayloads = answerCodecs.map(c => c.payload);
  
  const commonCodecs = offerPayloads.filter(payload => answerPayloads.includes(payload));
  
  if (commonCodecs.length === 0) {
    return { status: "MISMATCH", message: "No common codecs found", severity: "CRITICAL" };
  }
  
  // Check if high-quality codecs are available but not selected
  const hdCodecs = ["111", "112", "113", "114"]; // Opus payloads
  const hasHdOffer = offerPayloads.some(p => hdCodecs.includes(p));
  const hasHdAnswer = answerPayloads.some(p => hdCodecs.includes(p));
  
  if (hasHdOffer && !hasHdAnswer) {
    return { status: "DEGRADED", message: "HD codec offered but not selected", severity: "WARNING" };
  }
  
  return { status: "OK", message: "Codec negotiation successful", severity: "INFO" };
};

The Trap: Ignoring SDP Attributes
Codecs are not just about payload types. SDP attributes like a=fmtp (format parameters) can also cause issues. For example, G.729 requires specific frame lengths. If the provider sends an a=fmtp line with an unsupported frame length, the codec will fail even if the payload type matches.

Architectural Decision:
Extend your SDP parser to extract a=fmtp parameters. Validate these parameters against known standards for each codec. This adds complexity but significantly increases the accuracy of your diagnostics.

Validation, Edge Cases & Troubleshooting

Edge Case 1: SIP Re-INVITEs During Call Transfer

The Failure Condition:
A user transfers a call. The ladder diagram shows a new INVITE in the middle of the call. The media path changes, but the diagram does not clearly indicate the new media endpoints.

The Root Cause:
SIP re-INVITEs are used to update session parameters, such as changing the media destination during a transfer. If your diagram treats all INVITEs as initial calls, you will lose context of the media path change.

The Solution:
Group SIP messages by dialogId. A re-INVITE belongs to the same dialog as the initial INVITE. In your ladder diagram, draw a dashed line or a label indicating “Re-INVITE” to distinguish it from the initial setup. Update the media endpoints in the diagram to reflect the new c= (connection) and m= (media) addresses in the re-INVITE.

Edge Case 2: TLS Certificate Errors on Direct Trunks

The Failure Condition:
Calls fail with a 503 Service Unavailable or a TLS handshake error. The SIP trace shows no 100 Trying response.

The Root Cause:
The Genesys Cloud edge cannot establish a TLS connection with the provider due to an expired or invalid certificate on the provider’s side.

The Solution:
Check the TLS handshake logs in the Genesys Cloud admin console under Telephony > Trunks > TLS Settings. Ensure the provider’s certificate is in the trusted certificate store. In your ladder diagram, if no SIP messages are received after the INVITE, display a “TLS Handshake Failed” error banner.

Edge Case 3: Asymmetric Media Paths

The Failure Condition:
The call connects, but one side has no audio. The SIP trace shows successful codec negotiation.

The Root Cause:
The SDP offer and answer show different IP addresses for the media path. This is common when NAT is involved. The provider sends media to the public IP, but the Genesys Cloud edge expects media on the private IP.

The Solution:
Compare the c= (connection) address in the SDP offer vs. the answer. If they differ, highlight the IP addresses in the ladder diagram. Check the NAT Traversal settings on the Genesys Cloud trunk. Ensure that Symmetric RTP is enabled if the provider supports it. This forces Genesys Cloud to send media from the same IP it receives media on, resolving asymmetry.

Official References