Skip to main content

​
Telnyx Calling — Documentation Index

TeXML, SIP trunking, UAC connections, WebRTC, and trusted calling. One section of the Telnyx developer docs (https://developers.telnyx.com). Root index: https://developers.telnyx.com/llms.txt · Full content for this section: https://developers.telnyx.com/development/llms/calling-llms-full-txt.md

​
Subsections

Focused per-subsection files (index + full content):

​
Voice API

  • Getting Started: Complete step-by-step guide to get started with Telnyx Voice API, from account creation to making your first call.
  • Commands and Resources: Reference of Telnyx Voice API commands and resources — calls, conferences, recordings, transcriptions — with usage patterns and lifecycle examples.
  • Webhooks: Reference for Voice API webhook events, payloads, configuration, and delivery behavior.
  • Sending Commands: Send voice commands to Telnyx Programmable Voice — answer, hangup, transfer, play audio, gather DTMF — with code examples and parameter reference.
  • Receiving Webhooks: Receive Telnyx Programmable Voice webhooks to drive call flow in real time. Includes event types, payload structure, retries, and SDK examples.
  • Command Retries: Telnyx’s Programmable Voice Resources page explain in detail how to
  • Speech-to-Text: Transcribe Telnyx Programmable Voice calls with the Voice API and TeXML. Includes language selection, real-time vs post-call modes, and code samples.
  • Text-to-Speech: Use text-to-speech on Telnyx Programmable Voice to speak prompts and dynamic content. Supports AWS Polly, Google, ElevenLabs, Azure, and Telnyx voices.
  • Call Queueing: how to use the Telnyx Call Queue API to create and manage call
  • Answering Machine Detection: Telnyx’s Programmable Voice lets you build custom answering machine
  • Call Recordings Storage: Telnyx’s Programmable Voice gives customers the ability to store
  • Pay: Accept payments securely over the phone. Telnyx’s Pay over Voice feature guides callers through an automated IVR flow to collect payment details, sends them to the configured payment processor, and r…
  • SIPREC Server Configuration: Configure a Telnyx SIPREC session recording server (SRS) to capture and store VoIP call media for compliance, quality assurance, and call archival.
  • SIPREC Client: Use the Telnyx SIPREC client to record Voice API and TeXML calls to an external SIPREC server. Includes session setup, metadata, and security options.
  • Gather Using AI: Capture caller input with AI on Telnyx Programmable Voice. Use Gather using AI to transcribe speech, classify intent, and branch your IVR dynamically.
  • Attach an AI Assistant to a Call: Learn how to attach a pre-configured AI assistant to a live call using the ai_assistant_start command.
  • Dialogflow Integration: Telnyx’s Dialogflow tutorial will teach you how to create and
  • Real-Time Media Streaming: Stream call audio in real time over WebSockets with Telnyx Programmable Voice for transcription, AI integration, and bidirectional media.
  • Conversation Relay: Use Conversation Relay to connect Telnyx calls to your WebSocket application for real-time speech, DTMF, and text-to-speech control.
  • Noise Suppression: Enable AI noise suppression on Telnyx Programmable Voice calls to remove background noise from inbound and outbound audio for clearer conversations.
  • Deepfake Detection: Detect AI-generated voices on live calls with Telnyx’s real-time
  • IVR: How to build an IVR using Telnyx Voice API. Start building on Telnyx today.
  • Call Center: How to build a call center application using Telnyx APIs. Start
  • Call Tracking: How to build a call tracking application using Telnyx Voice API.
  • Conferencing: How to build a conferencing using Telnyx Voice API. Start building
  • Voice API Services in Europe: Telnyx Voice API services in Europe — use European points of presence to reduce latency and meet data residency requirements for EU-based voice apps.
  • L1 Account Restrictions: Restrictions that apply to Programmable Voice on Telnyx L1 verified accounts — automated speak warnings, call limits, and steps to upgrade verification.

​
TeXML

  • TeXML Fundamentals: Learn how TeXML, an XML-based markup language, defines call control and processing instructions for programmable voice applications on Telnyx.
  • Instruction Fetching: How Telnyx fetches TeXML instructions for inbound and outbound calls
  • TeXML Interpreter: TeXML is an XML based data structure used by Telnyx to build quick applications
  • TeXML TwiML Compatibility: Side-by-side comparison of TeXML and TwiML verb and noun support, helping you migrate Twilio call control instructions to Telnyx with minimal changes.
  • Dynamic TeXML Bins: Generate dynamic TeXML responses with Telnyx TeXML Bins. Inject runtime variables into TwiML-compatible call flows without standing up a backend.
  • Sending HTTP Requests: Send outbound HTTP requests from TeXML on Telnyx Programmable Voice. Use HttpRequest verbs to call your APIs mid-conversation with code samples.
  • TeXML Answering Machine: Learn how to enable and configure Answering Machine Detection (AMD)
  • Dial
  • Say
  • Play
  • Gather
  • AIGather
  • AIAssistant
  • Record
  • Conference
  • Enqueue
  • Hangup
  • Pause
  • Redirect
  • Reject
  • Stream
  • ConversationRelay
  • Connect
  • HttpRequest
  • Leave
  • Refer
  • Siprec
  • Start
  • Stop
  • Suppression
  • Transcription
  • Applications: Manage TeXML applications for handling inbound and outbound calls.
  • Calls: Initiate outbound calls and manage individual call legs programmatically through the Telnyx TeXML REST API, including state, metadata, and lifecycle.
  • Conferences: Create, modify, and end multi-participant audio conferences with the Telnyx TeXML REST API, including participant management and recording controls.
  • Conference participants: Add, mute, hold, and remove individual call legs from TeXML conferences using the Telnyx Conference Participants REST API for fine-grained control.
  • Recordings: List, retrieve, and manage audio recordings produced from TeXML calls and conferences, including metadata like format, duration, and storage URLs.
  • Transcripts: Retrieve and manage transcription results produced from TeXML calls when the Transcription verb is executed or transcription is enabled on a call.
  • SIPREC: Start and manage SIPREC media recording sessions for TeXML calls, delivering call audio to external SIP recording servers for compliance use cases.
  • Streams: Start, stop, and manage media streaming sessions over WebSockets for TeXML calls to deliver real-time audio to external transcription or AI systems.
  • Queues: Create and manage Telnyx TeXML queues to hold waiting callers in FIFO order before connecting them to agents, with hold music and announcement support.
  • TeXML application secrets: Securely store, manage, and retrieve credentials, API keys, and tokens used by TeXML applications via the Telnyx TeXML Secrets REST API.

​
SIP Trunking

  • Overview: Overview of Telnyx SIP Trunking — buy numbers, create a SIP connection, configure authentication and routing, and place your first inbound/outbound call.
  • Overview: SIP authentication methods supported by Telnyx SIP Trunking — credential, IP, FQDN, and token authentication. Compare and choose the right method.
  • IP Authentication Token: Use IP authentication tokens on Telnyx SIP Trunking to authenticate calls without a fixed allowlist. Generate tokens and include them in SIP INVITEs.
  • Tech Prefix: Authenticate Telnyx SIP Trunking calls with a tech prefix. Prepend a numeric prefix to the request URI to identify your account on outbound INVITEs.
  • Caller ID Policy: Configure caller ID policy on Telnyx SIP Trunking connections. Allow, override, or block From and P-Asserted-Identity values on outbound calls.
  • Concurrent Call Limits: Set concurrent call limits on Telnyx SIP Trunking connections. Cap inbound and outbound channels per connection to protect capacity and control spend.
  • P-Charge-Info: Use the P-Charge-Info SIP header on Telnyx SIP Trunking to specify the billed number for outbound calls. Includes header format and behavior rules.
  • Outbound Voice Profiles for SIP trunking: Attach Outbound Voice Profiles to Telnyx SIP Trunking connections to control allowed destinations, rate limits, daily spend, and outbound routing.
  • Round Robin Routing: Distribute Telnyx SIP Trunking traffic evenly across multiple destinations using round-robin routing. Spread load between PBXes or geographic regions.
  • AnchorSite: Configure AnchorSite to control which Telnyx PoP handles media routing for SIP calls, with latency-based and manual selection modes.
  • Failover & Retries: Configure failover and retry behavior on Telnyx SIP Trunking. Route calls to backup destinations on SIP failure codes for high availability.
  • IP Whitelisting: Whitelist your IP addresses on Telnyx SIP Trunking connections to authenticate calls without credentials. Includes allowed CIDR ranges, edge IPs, and AI services IPs.
  • SRV Records: Use SRV records to route SIP traffic to Telnyx with automatic failover. Configure DNS for sip._udp / sip._tcp and let your PBX resolve Telnyx edges.
  • STUN/TURN: Connect WebRTC and NAT-traversed SIP clients to Telnyx using STUN and TURN servers. Includes hostnames, ports, and authentication for media relays.
  • Noise Suppression: Enable AI noise suppression on Telnyx SIP Trunking calls. Remove background noise from inbound or outbound audio for clearer phone conversations.
  • Jitter Buffer: Configure the adaptive jitter buffer on Telnyx SIP Trunking to smooth out RTP packet delay and improve voice quality on lossy or variable networks.
  • External Transfers: Use SIP REFER to transfer Telnyx SIP Trunking calls to external numbers. Configure your PBX or Voice API to hand off calls without bridging media.
  • SIP URI Calling: Place and receive SIP URI calls over Telnyx SIP Trunking. Dial endpoints by sip:user@domain instead of E.164 to connect VoIP systems directly.
  • Attestation Behavior: How Telnyx assigns STIR/SHAKEN attestation levels (A, B, C) to outbound calls. Understand the criteria, signed identity headers, and downstream impact.
  • Hosted Certificate: Use a Telnyx hosted STIR/SHAKEN certificate to sign outbound calls without managing your own CA. Includes setup, certificate rotation, and verification.
  • SIP Header Parameters: Reference of STIR/SHAKEN SIP header parameters used by Telnyx — Identity, attest, origid, dest, and how to read and verify a signed identity header.
  • Dynamic E911: Provision Dynamic E911 on Telnyx SIP Trunking for VoIP emergency calling. Register caller locations dynamically and route 911 calls to the correct PSAP.
  • Response Codes: Reference of SIP response codes you’ll see on Telnyx SIP Trunking calls — 4xx client errors, 5xx server errors, 6xx global failures, and what they mean.
  • Configuration Overview: Telnyx’s SIP trunking guide will help you configure your sip
  • LiveKit: Integrate Telnyx SIP Trunks with LiveKit for real-time audio and video applications. Includes Telnyx configuration and LiveKit SIP gateway setup steps.

​
UAC Connections

  • UAC Connections: SIP Attach connects your existing PBX or SIP platform to Telnyx. Create a UAC Connection so Telnyx registers to your PBX and routes calls to Telnyx applications.

​
WebRTC

  • Fundamentals: Core concepts for building real-time voice apps with the Telnyx Voice WebRTC SDKs — sessions, media negotiation, codecs, and call lifecycle basics.
  • Architecture: Architecture overview of the Telnyx Voice WebRTC SDKs covering signaling, media flow, ICE/STUN/TURN, and how clients connect to the Telnyx network.
  • SDK Commonalities: Concepts shared across the Telnyx Voice WebRTC SDKs (JS, iOS, Android, Flutter) — clients, calls, events, push, and authentication flows in one place.
  • Credential Connections: Authenticate the Telnyx Voice WebRTC SDKs with credential-based SIP connections. Set up SIP credentials and login flow for browser and mobile apps.
  • Telephony Credentials: Authenticate the Telnyx Voice WebRTC SDKs with telephony credentials. Create per-user logins tied to a SIP connection for browser and mobile apps.
  • JWTs: Authenticate the Telnyx Voice WebRTC SDKs with short-lived JWT tokens. Generate, refresh, and validate tokens for secure browser and mobile clients.
  • Overview: How push notifications work with Telnyx WebRTC SDKs to deliver incoming call alerts when your app is backgrounded or terminated.
  • Android: Set up Firebase Cloud Messaging and configure the Telnyx Portal to receive push notifications for incoming calls on Android.
  • iOS: Configure APNS VoIP push certificates and the Telnyx Portal to receive push notifications for incoming calls on iOS.
  • Flutter: Configure push notifications for the Telnyx Flutter Voice SDK on both Android and iOS.
  • React Native: Configure push notifications for the Telnyx React Native Voice SDK on both Android and iOS.
  • JS SDK Demo App: Run the Telnyx JavaScript WebRTC demo app to make and receive browser calls, test hold, mute, and transfer, and inspect the open-source code.
  • JS SDK Anatomy: Anatomy of the Telnyx JavaScript Voice SDK — the TelnyxRTC client, Call objects, and how events flow between your web app and the Telnyx network.
  • Contact Center (CCaaS): Build a complete contact center solution using Telnyx WebRTC with advanced call control and routing capabilities.
  • Outbound Dialer: Build an automated outbound dialer system using Telnyx WebRTC and Call Control API for high-volume calling campaigns.
  • Call Detail Records: Use WebRTC call detail records (CDRs) on Telnyx to diagnose Voice SDK problems — connection failures, audio quality issues, and session lifecycle events.
  • Debug Logs: Enable and read debug logs from the Telnyx Voice WebRTC SDKs to diagnose signaling, media, and call quality issues in production and development.
  • Interpreting Debug Data: Step-by-step guide to interpreting debug logs from the Telnyx Voice WebRTC SDKs — read INVITE/BYE flows, ICE candidates, and identify common errors.
  • WebRTC JS SDK quickstart: Get started with the Telnyx WebRTC JS SDK in 5 minutes. Make and receive calls from your browser.
  • Build a Call Center Agent: Step-by-step tutorial for building a call center agent with the Telnyx WebRTC JS SDK — answering calls, muting, and holding.
  • WebRTC JS SDK authentication: How to authenticate the Telnyx WebRTC JS SDK using JWT, credentials, or anonymous login. Includes token refresh and security best practices.
  • Network Connectivity Requirements: Required endpoints, ports, and firewall configuration for the Telnyx WebRTC JS SDK.
  • Device Management: How to select, switch, and manage audio input/output devices (microphones, speakers) with the Telnyx WebRTC JS SDK.
  • Manage Multiple Calls: How to handle concurrent calls, call waiting, and per-call media element management with the Telnyx WebRTC JS SDK.
  • Call Report Stats: How to enable, access, and interpret WebRTC call reports from the Telnyx JS SDK — RTT, jitter, packet loss, ICE data, and audio quality metrics.
  • Reconnection & Call Recovery: How the Telnyx WebRTC JS SDK handles reconnection, when calls survive network interruptions, and how to configure recovery behavior.
  • Framework Integration: How to integrate the Telnyx WebRTC JS SDK with React, Next.js, Vue, and other JavaScript frameworks.
  • Debug Data & Call Quality Analysis: How to collect, interpret, and troubleshoot WebRTC call quality using debug reports, call reports, and the Telnyx debug visualizer.
  • WebRTC JS SDK error handling: Handle errors, warnings, and connection failures with the Telnyx WebRTC JS SDK, including fatal errors, recommended recovery actions, reconnection behavior, and version-specific guidance.
  • Production Best Practices: Production deployment guide for the Telnyx WebRTC JS SDK — security, reliability, performance, and monitoring.
  • TelnyxRTC Class: Main entry point for the Telnyx WebRTC JS SDK. Connect, disconnect, and manage calls.
  • Call Class: The Call object represents an active voice call — make, answer, hang up, mute, and hold.
  • IClientOptions: Configuration options for the TelnyxRTC client — authentication, connection, ICE, call reports, and debugging.
  • ICallOptions: Configuration options for placing a call with the Telnyx WebRTC JS SDK.
  • INotification: Notification object emitted by the Telnyx WebRTC JS SDK for call updates, media events, and SDK notifications.
  • Switch/Server Events: Events emitted by the Telnyx signaling server during the WebRTC call lifecycle — login, invite, answer, bye, and more.
  • How WebRTC Signaling Works: How the Telnyx WebRTC JS SDK communicates with Telnyx infrastructure — WebSocket connection, SIP signaling, and the full call flow from dial to hangup.
  • How ICE & TURN Work: How ICE connectivity checks, STUN, and TURN relay work in the Telnyx WebRTC JS SDK — and why they matter for call quality.
  • Authentication Architecture: How the three Telnyx WebRTC authentication methods relate — Credential Connections, Telephony Credentials, and JWTs — and when to use each.
  • Call State Lifecycle: Every state a Telnyx WebRTC call goes through, what triggers each transition, and what your app should do at each stage.
  • WebRTC JS ChangeLog: Release notes for the Telnyx JavaScript Voice SDK (@telnyx/webrtc). Track new features, breaking changes, bug fixes, and deprecations by version.
  • React quickstart: Get started with the Telnyx React SDK for building WebRTC voice applications in React.
  • Voice native iOS Client SDK: Telnyx iOS Voice Client SDK — add real-time SIP voice and video calling to native iOS apps with CallKit integration, VoIP push notifications, and Swift support.
  • WebRTC iOS Client: TxClient reference for the iOS Voice SDK. Main entry point used to log in, place outbound calls, and receive inbound calls in Swift and SwiftUI apps.
  • WebRTC iOS Call: Call class reference for the Telnyx iOS Voice SDK. Represents an active audio or video call and exposes accept, decline, hold, mute, and DTMF methods.
  • WebRTC iOS Call: Call extensions in the Telnyx iOS Voice SDK. Convenience methods added to the Call class for CallKit integration, state queries, and audio routing.
  • WebRTC iOS Client: TxClient extensions for the iOS Voice SDK. Helper methods added to the TelnyxRTC client for state queries, CallKit integration, and convenience APIs.
  • WebRTC iOS Call State: CallState enum reference for the iOS Voice SDK. Represents the current state of a call (NEW, CONNECTING, RINGING, ACTIVE, DONE) and supported transitions.
  • WebRTC iOS TxClientDelegate: TxClientDelegate protocol reference for the iOS Voice SDK. Implement these methods to receive call state changes, socket events, and backend updates.
  • WebRTC iOS Call Info: TxCallInfo struct reference for the iOS Voice SDK. Contains information about the current call — callerName, callerNumber, callId, and direction.
  • WebRTC iOS Client Configuration: TxConfig struct reference for the iOS Voice SDK. Structure used to configure TelnyxRTC client logins, ringtone, ringback tone, logging, and push.
  • WebRTC iOS Client Push Notifications Configuration: TxPushConfig struct reference for the iOS Voice SDK. Properties used to configure push notification credentials and behavior for incoming calls.
  • WebRTC iOS Client Push IP Notifications Configuration: TxPushIPConfig struct reference for the iOS Voice SDK. Properties used to configure the server endpoint for VoIP push notification delivery.
  • iOS Portal Setup: Configure VoIP push credentials in the Telnyx portal for the iOS Voice SDK. Upload your APNs VoIP certificate so Telnyx can deliver incoming call pushes.
  • iOS Push Notification Setup: Set up VoIP push notifications in your iOS app to wake the Telnyx Voice SDK on incoming calls. Covers PushKit, CallKit reporting, and SDK integration.
  • Troubleshooting: Troubleshoot iOS Voice SDK push notification problems — APNs VoIP delivery, PushKit registration, CallKit reporting timing, and missed call symptoms.
  • WebRTC iOS SDK AI Voice Assistant Introduction: Introduction to building AI-powered voice assistants with the iOS Voice SDK
  • WebRTC iOS SDK AI Voice Assistant Anonymous Login: Learn how to implement anonymous login for AI voice assistants with the iOS Voice SDK
  • WebRTC iOS SDK AI Voice Assistant Starting Conversations: Learn how to start conversations with AI voice assistants using the iOS Voice SDK
  • WebRTC iOS SDK AI Voice Assistant Text Messaging: Learn how to implement text messaging with AI voice assistants using the iOS Voice SDK
  • WebRTC iOS SDK AI Voice Assistant Transcript Updates: Learn how to handle transcript updates with AI voice assistants using the iOS Voice SDK
  • WebRTC Stats: WebRTC stats changelog for the Telnyx iOS Voice SDK. Updates and improvements to call-quality metrics, packet stats, and diagnostic data exposed by the SDK.
  • WebRTC iOS SDK Error Handling: Handle errors in the Telnyx iOS Voice SDK. Reference of TxError types, socket error codes, and patterns for surfacing call failures in your iOS app.
  • WebRTC iOS ChangeLog: Release notes for the Telnyx iOS Voice SDK. Track new features, breaking changes, bug fixes, and deprecations across published Swift package versions.
  • WebRTC Android Quickstart: Quickstart for the Telnyx Android Voice SDK using the telnyx_common drop-in module. Add SIP voice calling to your Android app in under 10 minutes.
  • Android Voice Client SDK: Telnyx Android Voice Client SDK — add real-time SIP audio and video calling to native Android apps, with push notifications and a drop-in module.
  • WebRTC Android Client: TelnyxClient class reference for the Telnyx Android Voice SDK. The main entry point — log in, connect, place outbound calls, and receive incoming calls.
  • WebRTC Android Call: Call class reference for the Telnyx Android Voice SDK. Represents an active audio or video call and exposes accept, decline, hold, mute, and DTMF methods.
  • WebRTC Android Config: TxConfig reference for the Telnyx Android Voice SDK. Defines login credentials, ringtone, logging level, and notification settings for TelnyxClient.
  • WebRTC Android ReceivedMessageBody: ReceivedMessageBody reference for the Android Voice SDK. Data class describing the structure of every message received over the Telnyx socket connection.
  • WebRTC Android SocketResponse: SocketResponse reference for the Android Voice SDK. Data class used for communication between the Telnyx WebSocket connection and TelnyxClient.
  • Android Portal Setup: Configure push notification credentials in the Telnyx portal for the Android Voice SDK. Upload your FCM server key so Telnyx can deliver call pushes.
  • Notification Quickstart for Android: Quickstart for adding push notifications to your Android Voice SDK app using the telnyx_common module. Drop-in solution for incoming call wakeups.
  • Android Push Notification Setup: Set up push notifications in your Android app to wake the Telnyx Voice SDK on incoming calls. Covers Firebase Cloud Messaging integration and SDK setup.
  • Android Push Troubleshooting: Troubleshoot push notification problems with the Android Voice SDK — debug FCM delivery, wakeup issues, and missing calls in foreground and background.
  • WebRTC Android SDK AI Voice Assistant Introduction: Introduction to building AI-powered voice assistants with the Android Voice SDK
  • WebRTC Android SDK AI Voice Assistant Anonymous Login: Learn how to implement anonymous login for AI voice assistants with the Android Voice SDK
  • WebRTC Android SDK AI Voice Assistant Starting Conversations: Learn how to start conversations with AI voice assistants using the Android Voice SDK
  • WebRTC Android SDK AI Voice Assistant Text Messaging: Learn how to implement text messaging with AI voice assistants using the Android Voice SDK
  • WebRTC Android SDK AI Voice Assistant Transcript Updates: Learn how to handle transcript updates with AI voice assistants using the Android Voice SDK
  • WebRTC Stats: WebRTC stats changelog for the Telnyx Android Voice SDK. Updates and improvements to call-quality metrics, packet stats, and diagnostic data exposed by the SDK.
  • WebRTC Call Reports: Collect and send WebRTC call statistics for troubleshooting and monitoring
  • WebRTC Android SDK Error Handling: Error handling reference for the Telnyx Android Voice SDK. List of common errors, TelnyxSocketError codes, and patterns for surfacing call failures.
  • WebRTC Android ChangeLog: Release notes for the Telnyx Android Voice SDK. Track new features, breaking changes, bug fixes, and deprecations across published versions.
  • React Native SDK: Official Telnyx React Native SDK for building cross-platform real-time communication applications with WebRTC.
  • React Native SDK Quickstart: Get started with the Telnyx React Native SDK in minutes. Complete setup guide from installation to making your first call.
  • Call: Represents a call with reactive state streams. Wraps the underlying Telnyx Call object and provides reactive streams for all call state changes.
  • TelnyxVoipClient: Main VoIP client class for React Native applications. Provides methods for call management, user registration, and WebRTC functionality.
  • CredentialConfig: Configuration for credential-based authentication with SIP username and password.
  • TokenConfig: Reference for the TokenConfig interface used to configure JWT token-based authentication when initializing the Telnyx WebRTC React Native SDK.
  • TelnyxVoipClientOptions: Configuration options for TelnyxVoipClient including app state management and debug settings.
  • TelnyxVoiceAppOptions: Configuration options for the TelnyxVoiceApp component including VoIP client management and push notification handling.
  • Portal Setup: Configure push notifications for React Native applications in the Telnyx Portal.
  • App Setup: Configure your React Native application for push notifications on both iOS and Android platforms.
  • Error Handling: Common errors, debugging techniques, and troubleshooting tips for the Telnyx WebRTC React Native SDK, including connection failures and call-state errors.
  • Changelog: Release notes and version history for the Telnyx WebRTC React Native SDK, including new features, breaking changes, bug fixes, and upgrade guidance.
  • Flutter Voice Client SDK: Telnyx Flutter Voice SDK — add real-time SIP voice and video calling to Flutter and Dart applications with push notifications, CallKit, and ConnectionService.
  • WebRTC Flutter Client: TelnyxRTC client reference for the Flutter Voice SDK. Main entry point used to log in, place outbound calls, and receive inbound calls in Dart and Flutter apps.
  • WebRTC Flutter Call: Call class reference for the Telnyx Flutter Voice SDK. Represents an active audio or video call and exposes accept, decline, hold, mute, and DTMF methods.
  • Flutter WebRTC SDK Message: TelnyxMessage reference for the Flutter Voice SDK. Represents an inbound message received from the Telnyx socket — events, payload fields, and parsing.
  • Flutter WebRTC SDK Socket Error Message: TelnyxSocketError reference for the Flutter Voice SDK. Represents an error received over the Telnyx WebSocket — codes, messages, and recovery patterns.
  • WebRTC Flutter Call State: CallState enum reference for the Flutter Voice SDK. Represents the current state of a call (NEW, CONNECTING, RINGING, ACTIVE, DONE) and supported transitions.
  • Flutter WebRTC SDK Socket Message Handler: onSocketMessageReceived event handler for the Flutter Voice SDK. Callback invoked when messages are received from the Telnyx WebSocket connection.
  • Flutter WebRTC SDK Socket Error Handler: onSocketErrorReceived event handler for the Flutter Voice SDK. Callback invoked when errors are received from the Telnyx WebSocket connection.
  • WebRTC Flutter Client Configuration: Config method object for the Flutter Voice SDK. Defines parameters used to configure the TelnyxClient login — credentials, ringtone, logging, and more.
  • WebRTC Flutter Incoming Invite Object: IncomingInviteParams method object for the Flutter Voice SDK. Information about an incoming call that can be answered, declined, or rejected by the app.
  • WebRTC Flutter Client Push Notifications Configuration: PushMetadata reference for the Flutter Voice SDK. Properties carried in push notification payloads to wake the SDK and reconcile incoming calls.
  • Flutter Push Notification Portal Setup: Configure push notification credentials in the Telnyx portal for the Flutter Voice SDK. Upload FCM and APNs/VoIP credentials for incoming call delivery.
  • Flutter Push Notification App Setup: Set up push notifications in your Flutter app to wake the Telnyx Voice SDK on incoming calls. Covers FCM, APNs/VoIP, and platform-channel wiring.
  • Flutter Push Troubleshooting: Troubleshoot push notification problems with the Flutter Voice SDK — debug FCM and VoIP push delivery, missed calls, and platform-channel issues.
  • WebRTC Flutter SDK AI Voice Assistant Introduction: Introduction to building AI-powered voice assistants with the Flutter Voice SDK
  • WebRTC Flutter SDK AI Voice Assistant Anonymous Login: Learn how to implement anonymous login for AI voice assistants with the Flutter Voice SDK
  • WebRTC Flutter SDK AI Voice Assistant Starting Conversations: Learn how to start conversations with AI voice assistants using the Flutter Voice SDK
  • WebRTC Flutter SDK AI Voice Assistant Text Messaging: Learn how to implement text messaging with AI voice assistants using the Flutter Voice SDK
  • WebRTC Flutter SDK AI Voice Assistant Transcript Updates: Learn how to handle transcript updates with AI voice assistants using the Flutter Voice SDK
  • WebRTC Stats: WebRTC stats changelog for the Telnyx Flutter Voice SDK. Updates and improvements to call-quality metrics, packet stats, and diagnostic data exposed by the SDK.
  • WebRTC Flutter SDK Error Handling: Error handling reference for the Telnyx Flutter Voice SDK. Common errors, socket error codes, and patterns for surfacing call failures in Flutter apps.
  • WebRTC Flutter ChangeLog: Release notes for the Telnyx Flutter Voice SDK. Track new features, breaking changes, bug fixes, and deprecations across published Dart package versions.

​
Trusted Calling

  • Overview: Display your business name, logo, and call reason on recipients’ phone screens with Branded Calling.
  • Quickstart: Get your first branded call working, from Terms of Service to a verified DIR with phone numbers attached.
  • Pricing: Branded Calling pricing — enterprise registration is free, brand registration has a one-time and monthly cost, plus a per-call charge.
  • Overview: Register your business as an enterprise to use Branded Calling.
  • Overview: Create and manage Display Identity Records (DIRs), the display identity shown to call recipients.
  • Overview: Attach phone numbers to a verified DIR and inspect the carrier-network vetting batches.
  • Overview: Configure the call reason text displayed to recipients, explaining why your business is calling.
  • Overview: Accept the Branded Calling Terms of Service before activating Branded Calling on an enterprise.
  • Overview: Inspect, contest, and resolve infringement claims filed against your DIR.

​
API Reference (Calling)

​
Call Control Applications

​
Call Information

​
Debugging

  • List call events: Filters call events by given filter parameters. Events are ordered by occurred_at. If filter for leg_id or application_session_id is not present, it only…

​
Call Commands

  • Dial: Dial a number or SIP URI from a given connection. A successful response will include a call_leg_id which can be used to correlate the command with subsequent…
  • Answer call: Answer an incoming call. You must issue this command before executing subsequent commands on an incoming call.
  • Bridge calls: Bridge two call control calls.
  • Start AI Assistant: Start an AI assistant on the call.
  • Join AI Assistant Conversation: Add a participant to an existing AI assistant conversation. Use this command to bring an additional call leg into a running AI conversation.
  • Stop AI Assistant: Stop an AI assistant on the call.
  • Add messages to AI Assistant: Add messages to the conversation started by an AI assistant on the call.
  • Start Conversation Relay: Start a Conversation Relay session on an active call. Conversation Relay connects the call audio to your WebSocket so your application can exchange realtime me…
  • Stop Conversation Relay: Stop the active Conversation Relay session on a call.
  • Update client state: Updates client state
  • Enqueue call: Put the call in a queue.
  • Forking start: Call forking allows you to stream the media from a call to a specific target in realtime.
  • Forking stop: Stop forking a call.
  • Gather: Gather DTMF signals to build interactive menus.
  • Gather stop: Stop current gather.
  • Gather using AI: Gather parameters defined in the request payload using a voice assistant.
  • Gather using audio: Play an audio file on the call until the required DTMF signals are gathered to build interactive menus.
  • Gather using speak: Convert text to speech and play it on the call until the required DTMF signals are gathered to build interactive menus.
  • Hangup call: Hang up the call.
  • Reject a call: Reject an incoming call.
  • Remove call from a queue: Removes the call from a queue.
  • Play audio URL: Play an audio file on the call. If multiple play audio commands are issued consecutively,
  • Stop audio playback: Stop audio being played on the call.
  • Recording start: Start recording the call. Recording will stop on call hang-up, or can be initiated via the Stop Recording command.
  • Recording stop: Stop recording the call.
  • Record resume: Resume recording the call.
  • Record pause: Pause recording the call. Recording can be resumed via Resume recording command.
  • Send DTMF: Sends DTMF tones from this leg. DTMF tones will be heard by the other end of the call.
  • SIP Refer a call: Initiate a SIP Refer on a Call Control call. You can initiate a SIP Refer at any point in the duration of a call.
  • SIPREC start: Start siprec session to configured in SIPREC connector SRS.
  • SIPREC stop: Stop SIPREC session.
  • Speak text: Convert text to speech and play it back on the call. If multiple speak text commands are issued consecutively, the audio files will be placed in a queue awaiti…
  • Streaming start: Start streaming the media from a call to a specific WebSocket address or Dialogflow connection in near-realtime. Audio will be delivered as base64-encoded RTP…
  • Streaming stop: Stop streaming a call to a WebSocket.
  • Noise Suppression Start (BETA)
  • Noise Suppression Stop (BETA)
  • Switch supervisor role: Switch the supervisor role for a bridged call. This allows switching between different supervisor modes during an active call
  • Transcription start: Start real-time transcription. Transcription will stop on call hang-up, or can be initiated via the Transcription stop command.
  • Transcription stop: Stop real-time transcription.
  • Transfer call: Transfer a call to a new destination. If the transfer is unsuccessful, a call.hangup webhook for the other call (Leg B) will be sent indicating that the tran…

​
Callbacks

​
Conference Commands

​
TeXML Applications

​
TeXML REST Commands

​
Connections

  • List connections: Returns a list of your connections irrespective of type.
  • Retrieve a connection: Retrieves the high-level details of an existing connection. To retrieve specific authentication information, use the endpoint for the specific connection type.

​
Credential Connections

​
IPs

​
IP Connections

​
FQDNs

​
FQDN Connections

​
UAC Connections

  • List UAC connections: Returns a list of your UAC connections. A UAC (User Agent Client) Connection registers Telnyx to your PBX — the opposite of a standard SIP trunk, where the PBX…
  • Create a UAC connection: Creates a UAC connection. A UAC (User Agent Client) Connection registers Telnyx to your PBX — the opposite of a standard SIP trunk, where the PBX registers to…
  • Retrieve a UAC connection: Retrieves the details of an existing UAC connection.
  • Update a UAC connection: Updates settings of an existing UAC connection.
  • Delete a UAC connection: Deletes an existing UAC connection.
  • Check a UAC Connection Registration Status: Checks the registration status for a UAC connection (registration_status) as well as the timestamp for the last SIP registration event (`registration_status_…
  • SIP Registration Status: Returns the live SIP registration state of a UAC connection: whether it is currently registered, when it last registered, and the last response Telnyx received…

​
Outbound Voice Profiles

​
Dialogflow Integration

​
External Connections

​
Call Recordings

​
Media Storage API

​
Credentials

​
Access Tokens

​
Push Credentials

​
Voice SDK Stats

  • List Voice SDK call reports: Returns paginated raw call report stats JSON payloads stored for the authenticated user. The user is derived from Telnyx authentication, not from request param…
  • Retrieve Voice SDK call reports by call ID: Returns raw call report stats JSON payloads stored for the authenticated user and call_id. The user is derived from Telnyx authentication, not from request p…

​
Enterprises

  • List enterprises: Return the enterprises you own, paginated. The default page size is 20; the maximum is 250.
  • Create an enterprise: Create the legal entity (enterprise) that represents your business on the Telnyx platform.
  • Get an enterprise: Retrieve a single enterprise by id. Returns 404 if the id does not exist or does not belong to your account.
  • Replace an enterprise: Replace the enterprise’s mutable fields. Only mutable fields may be sent. Server-assigned and immutable fields (id, record_type, created_at, updated_at…
  • Delete an enterprise: Soft-delete an enterprise.
  • Activate Branded Calling on an enterprise: Branded Calling is a paid product that must be activated on each enterprise. Activation is idempotent:

​
Display Identity Records

  • List DIRs in an enterprise: Return the DIRs (Display Identity Records) belonging to a single enterprise. Pagination is JSON:API style (page[number], page[size], max 250). Supports `fi…
  • Create a Display Identity Record (DIR): Create a new DIR under the given enterprise. The DIR starts in draft status; it must be submitted (POST .../submit) and approved by Telnyx before any phone…
  • List all DIRs across your enterprises: Returns every DIR (Display Identity Record) you own, across all of your enterprises, as a single list. Pagination is JSON:API style (page[number], `page[size…
  • Get a DIR by id: Returns a single DIR by id. The enterprise is resolved server-side from the DIR id. Returns 404 if the DIR does not exist or is not yours.
  • Update a DIR: Edit a DIR. DIRs in draft, rejected, unsuccessful, or suspended can be edited freely: PATCH is a pure edit, status is never changed, and you re-vet b…
  • Delete a DIR: Delete a DIR. Failure modes: 400 if a child phone number is in a non-deletable status, 409 if the DIR has an unresolved infringement claim, 404 if the DI…
  • Submit a DIR for vetting: Submit a DIR for vetting. Sends the DIR back through the vetting cycle from any non-terminal status. When re-submitting from suspended or expired, the DIR’…
  • Render the Branded Calling LOA for a DIR: Generate a pre-filled Letter of Authorization (LOA) PDF for a DIR. Enterprise identity (legal name, DBA, address, contact, website, tax id) and the DIR display…

​
Reference Data

  • List supported DIR document types: Reference list of document_type values accepted by DirCreateRequest.documents[].document_type and the infringement-contest endpoint. Each entry has a stabl…
  • List standard call reasons: Telnyx maintains a library of pre-vetted call-reason phrases (e.g. “Appointment reminders”, “Billing inquiries”) that carry through DIR vetting smoothly. You c…
  • Validate a list of call reasons: Check up to 10 candidate call_reasons strings against Telnyx’s vetting heuristics before sending them on a DIR create or update. The endpoint flags strings t…

​
Comments

  • List comments on a DIR: List the comments on a DIR. The enterprise is resolved server-side from the DIR id.
  • Post a comment on a DIR: Post a customer comment on a DIR (for example, to respond to reviewer notes). Send only content (1–5000 chars) and an optional parent_comment_id; the serve…

​
Phone Numbers

  • List phone numbers attached to a DIR: List the phone numbers registered under a DIR. The enterprise is resolved server-side from the DIR id.
  • Add phone numbers to a DIR: Register phone numbers under a DIR. The enterprise is resolved server-side from the DIR id. Same body, failure modes, and batch semantics whichever path form y…
  • Remove phone numbers from a DIR: Deregister phone numbers from a DIR. The enterprise is resolved server-side from the DIR id. Returns a partial-success envelope.

​
Phone Number Batches

​
Infringement Claims

  • List infringement claims for a DIR: Return the trademark or copyright claims filed against this DIR. Each claim’s status is pending (newly filed; DIR auto-suspended), contested (you have su…
  • Update a DIR to resolve an infringement concern: Push a fix for a DIR that is suspended with an open infringement claim back into vetting. POST /dir/{dir_id}/submit is blocked while a claim is open, so th…
  • Get an infringement claim: Retrieve a single claim by id. Returns 404 if the claim does not exist or is not against a DIR you own.
  • Contest an infringement claim: Submit a written response and supporting documents disputing the claim. The first call moves the claim from pending to contested; subsequent calls append s…

​
Terms of Service