You already have a working voice stack. It can be a LiveKit app, a WebRTC pipeline, a Twilio Media Streams bridge, an Asterisk or FreeSWITCH deployment, or a bespoke WebSocket audio sender. A migration to ClutchCall Voice does not require a rewrite. The fastest and lowest-risk path is transport-first. Keep your runtime, your agent loop, your ASR/LLM/TTS chain, and your call control. Swap only the layer that moves audio. That layer becomes real-time media over QUIC/MoQT on the single :443 plane.

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The transport-first thesis

Many voice migrations stall because teams treat them as a full-platform port. This is not necessary. Your runtime already produces and consumes audio frames across a media boundary. That boundary is WebRTC’s SRTP, a WebSocket that carries PCM or µ-law, or RTP on a SIP trunk. In the migration, you re-point that one boundary at ClutchCall. Everything above it stays unchanged.

What you keep

You keep your conversation loop, VAD and turn-taking, prompts and tools, models and vendor credentials, call-control logic, and dashboards.

What you swap

You swap only the media transport. The media transport is the wire that carries encoded audio between your runtime and the endpoint (browser, app, or phone).
Whether ClutchCall also runs your agent is a separate, later choice. You can adopt the transport alone and keep your runtime indefinitely. See Keep Your Existing Agent Runtime. The menu below is only about the media boundary.

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Pick your path

Choose the path that matches where your audio lives today. Each path re-points one boundary. The detail page for each path describes the cutover.

From WebRTC

Keep the browser’s capture, echo cancellation, and Opus encoder. Divert the encoded frames onto QUIC/MoQT instead of ICE/DTLS/SRTP. This path is Shipped in the TypeScript SDK.

From Self-Hosted LiveKit

Install the transport plugin in your livekit-agents worker. Then retire the SFU. This path is Shipped. Or swap one import in a browser app. That path is Preview.

From Custom WebSocket Audio

Point a bespoke WebSocket audio sender at the engine’s WebSocket media fallback. Or connect your runtime to the raw publish/subscribe primitives.

From Twilio Media Streams

Move a µ-law Media Streams WebSocket onto a SIP PCMU trunk with zero transcode. Or bridge it as an external media vendor.

From Asterisk / FreeSWITCH

Re-point your SIP trunks and agent endpoints at ClutchCall’s built-in B2BUA, registrar, and dialplan. This path is Shipped. You do not need a separate SIP proxy.

Migration Checklist

Run this step-by-step cutover checklist after you pick a path.

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What each path swaps

Status shows what ships today. “Shipped” paths are proven end-to-end. “Preview” paths work for the core voice case. But they are not yet drop-in for every API of the source product. Read the callouts on each path page before you plan a cutover.
The two LiveKit rows are different products. Do not mix them up. The server-side transport plugin (livekit-plugins-clutchcall / @clutchcall/livekit-transport) is shipped. It installs into your agent worker and replaces the session’s audio IO, so there is no room to emulate. It has three limits: pcm16 audio, a register-and-wait worker (not scale-to-zero), and a need for outbound UDP/443.The browser livekit-compat shim is Preview. It covers the core voice path (publish/subscribe audio, data messages). It was tested browser-to-browser over the transport. But it is not yet a drop-in for every livekit-client API, and video and screenshare are deferred. Pin a version. Test your flows. See Keep Your Existing Agent Runtime.

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How a transport-first migration runs

Every path has the same shape. The per-source page has the detail. This section shows the sequence.
1

Provision your workspace

Onboarding creates your per-tenant endpoints: <workspace-id>.sip.clutchcall.dev for phone traffic and <workspace-id>.webrtc.clutchcall.dev for browser and app media. It also creates your relay at relay.clutchcall.dev. Nothing in your runtime changes yet.
2

Re-point one media boundary

Swap only the transport for the path you picked:
  • Divert encoded WebRTC frames onto QUIC/MoQT.
  • Or swap the LiveKit import.
  • Or point your WebSocket sender at the engine’s media fallback.
  • Or re-point a SIP trunk.
Your agent code stays unchanged.
3

Verify audio both ways

Check that uplink audio (caller/browser → runtime) flows. Check that downlink audio (runtime → caller) flows. Check quality. On telephony legs, the media plane scores MOS, jitter, and loss per call. Compare these scores against your old stack. See MOS, Jitter & Loss.
4

Cut over gradually

Route a slice of traffic to the new transport. A slice can be a test DID, a canary trunk, or a subset of browser sessions. Watch the dashboards. Then widen the slice. Follow the Migration Checklist for the full cutover sequence.

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Where the runtime line falls

Transport-first is one end of a spectrum. The minimal-change migration keeps your runtime and uses only the wire. You can also give the conversation loop to ClutchCall and bring only the models. That choice is a runtime adoption, not a transport swap. ClutchCall can run a cascaded ASR→LLM→TTS pipeline or a speech-to-speech provider for you.
LiveKit Agents and Pipecat have packaged plugins. Other stacks have the external-agent SDK or a bridge. For a livekit-agents worker, install livekit-plugins-clutchcall / @clutchcall/livekit-transport. For a Pipecat pipeline, install pipecat-telequick. In both cases, your agent runs over our transport directly. If you come from Twilio, Vapi, Plivo, or Telnyx, use the provider starters. The starters port each vendor’s official sample onto the telequick-agents external-agent SDK (see the integration pages). If you keep a vendor-hosted runtime in operation and put ClutchCall in front of it, that is a config-level bridge. See Keep Your Existing Agent Runtime.

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Next steps

Keep Your Existing Agent Runtime

Adopt the transport alone and leave your runtime in place. This page covers the shim and the raw primitives in depth.

WebRTC Diversion

The encoded-frame tap moves a browser WebRTC pipeline onto QUIC. It does not touch capture or the codec.

SIP Trunking

Re-point carrier trunks at the built-in B2BUA and registrar for telephony migrations.

Migration Checklist

Run the ordered cutover checklist after you choose a path.