Assigning Phone Numbers and WebRTC Configurations via the Telephony API

How-To-Articles
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Assigning Phone Numbers and WebRTC Configurations via the Telephony API

What This Guide Covers

This guide details the programmatic provisioning of PSTN and SIP trunk phone numbers, their assignment to user extensions or routing objects, and the configuration of WebRTC and Sofia dialing rules via the Genesys Cloud CX Telephony API. When complete, your deployment pipeline will automatically provision numbers, bind them to user endpoints, and enforce deterministic WebRTC media paths without manual UI intervention.

Prerequisites, Roles & Licensing

  • Licensing Tier: CX 1 or CX 2. WebRTC and Sofia endpoint capabilities are included in all CX tiers. Advanced routing or WEM add-ons are not required for telephony provisioning.
  • UI Permissions: Telephony > Phone Number > View, Telephony > Phone Number > Edit, Telephony > User Telephony Settings > Edit, Telephony > WebRTC Provider > View
  • OAuth Scopes: telephony:phone-number:view, telephony:phone-number:edit, telephony:user-settings:edit, telephony:web-r-t-c:view
  • External Dependencies: Active SIP Trunk or Genesys PSTN carrier integration, pre-provisioned user IDs in Genesys Cloud, valid OAuth 2.0 client credentials with service account authentication, and a network configuration object ID for WebRTC/Sofia endpoints.

The Implementation Deep-Dive

1. Programmatic Phone Number Provisioning and State Verification

Phone number provisioning in Genesys Cloud is an asynchronous operation. The API accepts the request, queues it for carrier validation, and returns a pending state immediately. Your orchestration logic must account for this delay to prevent race conditions during bulk onboarding.

Execute a POST request to the provisioning endpoint with the target number, provider mapping, and geographic location. The providerId must reference a pre-configured SIP Trunk or Genesys PSTN provider. Mismatched provider types trigger silent failures that surface hours later during call routing.

HTTP Method: POST
Endpoint: /api/v2/telephony/phone/numbers
JSON Payload:

{
  "type": "pstn",
  "number": "+14155552671",
  "providerId": "sip-trunk-acme-us-east",
  "locationId": "us_east_1",
  "routing": {
    "type": "queue",
    "id": "queue-inbound-sales",
    "displayName": "Sales Inbound Line"
  }
}

The Trap: Treating the 202 Accepted response as a confirmation of availability. The number enters a provisioning lifecycle state. If your pipeline immediately attempts to assign the number to a secondary routing object or triggers a WFM forecast sync, the API returns 409 Conflict because the telephony subsystem has not yet completed carrier registration.

Architectural Reasoning: We decouple provisioning from assignment to enforce idempotency. Your orchestration script must poll GET /api/v2/telephony/phone/numbers/{phoneNumberId} until the state field transitions to active. Implement exponential backoff starting at 3 seconds, capping at 60 seconds, with a maximum retry window of 300 seconds. This pattern prevents API throttling and aligns with carrier registration SLAs, which typically complete within 15 to 45 seconds for SIP trunks and up to 5 minutes for legacy PSTN carriers.

2. Binding Phone Numbers to Users and Preserving Routing Topology

Once the number reaches the active state, you must bind it to the correct telephony object. Genesys Cloud uses a single routing object per phone number. Updating this object replaces the entire routing configuration. You cannot append routing rules; you must reconstruct them.

Execute a PUT request to update the number assignment. Always fetch the current configuration first. Merge your new routing requirements with existing attributes. This prevents accidental detachment of IVR call flows or queue routing rules during CI/CD deployments.

HTTP Method: PUT
Endpoint: /api/v2/telephony/phone/numbers/{phoneNumberId}
JSON Payload:

{
  "routing": {
    "type": "user",
    "id": "user-agent-48291",
    "displayName": "Agent Direct Line",
    "routingType": "direct",
    "callflowId": "cf-inbound-ivr-main"
  }
}

The Trap: Overwriting the routing object without preserving the callflowId. When you assign a number directly to a user, Genesys Cloud bypasses the IVR unless you explicitly retain the callflowId in the payload. Dropping this field routes all inbound calls straight to the user extension, breaking compliance recordings, skills-based routing, and ACD analytics.

Architectural Reasoning: We treat routing objects as immutable snapshots. Your deployment pipeline should maintain a routing manifest in version control. Before issuing the PUT, your script reads the current manifest, applies the delta, and validates the merged object against the TelephonyPhoneNumber schema. This approach guarantees that IVR entry points, queue fallbacks, and after-hours routing remain intact during number reassignment. It also enables rollback capabilities without manual UI intervention.

3. Configuring WebRTC and Sofia User Settings with Network Determinism

WebRTC and Sofia endpoints rely on explicit network configuration to negotiate ICE candidates. Browser-based WebRTC does not auto-discover corporate STUN/TURN servers reliably. You must inject deterministic network profiles through the user telephony settings API.

Execute a PUT request to the user settings endpoint. The payload must include both webRtc and sofia objects, even if only one is active. Genesys Cloud evaluates both objects during endpoint registration. Missing one object causes the telephony subsystem to fall back to default network discovery, which fails in NAT-heavy or firewall-restricted environments.

HTTP Method: PUT
Endpoint: /api/v2/telephony/users/{userId}/settings
JSON Payload:

{
  "webRtc": {
    "enabled": true,
    "dialRules": [
      {
        "pattern": "^\\+1.*",
        "prefix": "9",
        "description": "Long distance prefix"
      },
      {
        "pattern": "^\\d{7,10}$",
        "prefix": "+1415",
        "description": "Local number normalization"
      }
    ],
    "networkConfigId": "web-rtc-nat-config-01",
    "codecs": ["G722", "OPUS", "PCMU"],
    "stunServers": ["stun:stun.l.google.com:19302", "stun:stun1.l.google.com:19302"]
  },
  "sofia": {
    "enabled": true,
    "networkConfigId": "sofia-nat-config-01",
    "codecs": ["G722", "PCMU", "PCMA"]
  }
}

The Trap: Omitting stunServers or relying solely on networkConfigId. The networkConfigId references a Genesys Cloud network profile, but it does not override browser-level ICE candidate generation. If you exclude explicit stunServers, the WebRTC endpoint will attempt to use public STUN servers that may be blocked by corporate firewalls. This results in ICE_FAILED states, one-way audio, or complete media handshake failure.

Architectural Reasoning: We enforce explicit STUN/TURN injection to guarantee media path predictability. Remote workforces operate across heterogeneous network topologies. By hardcoding STUN servers and defining precise dial rules, we eliminate browser negotiation variance. The dialRules array normalizes outbound dialing before SIP INVITE generation, preventing carrier rejection due to malformed E.164 formatting. We also prioritize G722 and OPUS codecs to ensure wideband audio quality, falling back to PCMU only for legacy gateway compatibility.

Validation, Edge Cases & Troubleshooting

Edge Case 1: Asynchronous Provisioning Timeout

  • The Failure Condition: The phone number remains in provisioning state beyond 300 seconds. Subsequent assignment attempts return 409 Conflict.
  • The Root Cause: The referenced providerId maps to a SIP trunk with disabled automatic number provisioning, or the carrier requires manual registration for the specific number range.
  • The Solution: Validate the providerType matches the type field in your payload. If using a SIP trunk, ensure autoProvisionNumbers is enabled in the trunk configuration. Implement a fallback polling strategy that checks GET /api/v2/telephony/phone/numbers/{id}/status for carrier rejection codes. Retry provisioning only after confirming the carrier accepts the number format.

Edge Case 2: WebRTC ICE Candidate Failure

  • The Failure Condition: Diagnostic logs show ICE_FAILED or CANDIDATE_GATHERING_TIMEOUT. Audio streams fail to establish despite successful SIP signaling.
  • The Root Cause: Corporate firewall blocks UDP ports 3478 through 3481, or the networkConfigId references a profile with restrictive NAT traversal settings.
  • The Solution: Inject a TURN relay server into the stunServers array using the turn: URI scheme. Verify that the network configuration object has forceTcp set to false unless UDP is explicitly blocked. Update the PUT /api/v2/telephony/users/{userId}/settings payload to include iceTransportPolicy: "all". Validate media path using GET /api/v2/telephony/diagnostics/web-r-t-c before rolling out to production users.

Edge Case 3: Routing Object Overwrite Cascade

  • The Failure Condition: After API deployment, inbound calls bypass the IVR and route directly to a queue or user. WEM recordings stop capturing IVR navigation paths.
  • The Root Cause: The PUT request replaced the routing object without preserving the callflowId. Genesys Cloud treats missing callflowId as a direct routing instruction.
  • The Solution: Implement a GET-merge-PUT pattern in your deployment script. Fetch the current routing configuration, apply only the target field delta, and re-serialize the complete object before issuing the PUT. Add a validation step that checks routing.callflowId !== null before committing. Reference the WFM skill group mapping guide to ensure routing changes do not break forecast-to-actual alignment.

Official References