New features
Video avatars — Render your agent as a photorealistic talking head during WebRTC calls. Choose between HeyGen and Tavus by settingavatar.provider on the agent config. Tavus delivers sub-600ms latency at 1080p; HeyGen streams alongside your existing voice pipeline. See Video Avatar for setup and field reference.OpenRouter LLM provider — Route LLM traffic through OpenRouter to access hundreds of models with automatic fallback routing. Configure primary and fallback models per agent to improve reliability when an upstream model is degraded. See providers.Updates
Pipecat 1.4 upgrade — The underlying voice pipeline has been upgraded from Pipecat 1.0 to 1.4 for improved stability and provider compatibility. No action required.Streaming audio and latency fixes — Audio resampling now drops empty buffers instead of pushing silence, reducing artifacts and lowering end-to-end latency on cascade pipelines.New features
Interactive API reference — You can now explore and test every TurnCall endpoint directly from the docs. The new API reference includes request and response schemas, example payloads, and a built-in playground.Open source under MIT license — TurnCall is now fully open source. The entire project is available under the MIT license, so you can self-host, fork, and contribute freely.Updates
Rebrand to TurnCall — The project has been renamed from Voicey to TurnCall. All API endpoints, configuration files, and documentation now use the TurnCall name consistently. No action is required if you are using the hosted API.New features
MCP server support for tools — You can now connect MCP servers to your agents for auto-discovered tool calling. Any tools exposed by your MCP server are automatically available during calls.Post-call analysis — TurnCall now automatically generates a structured post-call analysis after every call, including a summary, sentiment score, success evaluation, and custom data extraction. Thecall.ended webhook is enriched with the full transcript, recording URL, and analysis results.Agent versioning — Publish immutable agent versions, auto-promote phone numbers to the latest version, and roll back instantly when needed.A/B testing — Route traffic across agent versions with weighted A/B testing on phone numbers. Routing is deterministic by caller, so the same caller always reaches the same version.Cartesia STT/TTS provider — Cartesia is now available as a speech-to-text and text-to-speech provider, giving you another option for voice quality and latency tuning.Anthropic Claude as LLM provider — You can now use Anthropic Claude models as the LLM provider for your agents, alongside OpenAI and Ollama.Knowledge base with RAG — Upload documents to a knowledge base and attach it to agents. Three retrieval modes are available: prompt injection, automatic retrieval, and tool-based lookup.Pre-call init hook — Use the call-init server event to dynamically resolve agent configuration before the pipeline starts. You can also hand off mid-call between agents using the built-in handoff tool.Updates
Richer webhook payloads — Thecall.ended webhook now includes call metadata (from/to number, direction, duration), the full transcript, and the recording URL. All call events are dispatched to webhook subscribers.Call recording storage — Twilio call recordings are now automatically downloaded and stored locally or in S3.Bug fixes
Duplicatecall.started events — Fixed an issue where call.started was fired twice per call.Transcript sequencing — Transcript events now use database sequence numbers, preventing ordering collisions in high-throughput calls.Webhook payload format — Fixed the webhook event key to use event consistently (previously some payloads used event_type).New features
SMS and chat support — Agents can now handle text-based conversations over SMS and the Chat API. Sessions are managed automatically so returning users pick up where they left off.WhatsApp Business integration — Connect your agents to WhatsApp for both voice calls and text messages through the WhatsApp Business platform.Speech-to-speech mode — A new speech-to-speech pipeline delivers ultra-low-latency voice interactions powered by OpenAI Realtime and Gemini Live, bypassing the traditional STT → LLM → TTS chain.Bring Your Own Model (BYOM) — Point your agents at any OpenAI-compatible endpoint to use custom or self-hosted LLMs as the provider.WebRTC support — Launch browser-based voice calls directly from your web application without requiring a phone number.Smart Turn and voicemail detection — Improved turn-taking with Smart Turn V3 and Silero VAD reduces false interruptions. Incoming calls are now automatically screened for voicemail so your agent can hang up early instead of talking to a machine.Updates
Pipecat 1.0 migration — The underlying voice pipeline has been upgraded to Pipecat 1.0, improving stability and enabling new provider integrations.Renamed “assistant” to “agent” — All API endpoints and documentation now use “agent” consistently. The/v1/assistants endpoints have been replaced by /v1/agents.