to connect my agent with third party provider should i use websockets or sip

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both

Thanks — you want to support both phone calls (telephony) and real‑time AI model integration. Below is a simple, practical architecture and next steps you can follow to get both working together. High-level architecture (what to use) • Telephony/PSTN connectivity: SIP ◦ Use a SIP trunk from a provider (Twilio, Telnyx, Plivo, etc.) to handle inbound/outbound phone numbers and PSTN signaling/media. ◦ LiveKit will bridge SIP calls into a WebRTC room so your human or bot agent can join the call as a participant and speak with the caller. • Real‑time AI model interaction (assistant, transcription, TTS, NLU): WebSockets (or the model vendor’s realtime API) ◦ Use WebSockets between your agent backend and the AI model API for low‑latency audio/text streaming and inference. ◦ Keep the model connection separate from the SIP trunk; stream audio from the LiveKit room to the model via your backend and send model responses back into the room. How the pieces fit together (flow) 1. Caller dials PSTN number → SIP provider terminates call to your SIP trunk. 2. LiveKit receives SIP call and creates/joins a WebRTC room where the caller is a participant (caller media is RTP/SRTP). 3. Your agent (human or bot) joins the same LiveKit room as a participant (WebRTC). 4. For AI features: ◦ Your backend pulls audio from the LiveKit room (mix or single participant stream). ◦ Backend streams that audio over WebSockets to the AI model for transcription, NLU, or real‑time assistants. ◦ Model responses (text or generated audio) are sent back to the backend and injected into the LiveKit room (as TTS audio or text events) so the agent/caller can hear or see them. Key implementation considerations • Audio quality and transport: ◦ Keep SIP for signaling/media and WebRTC for in‑browser/agent media; don’t try to run telephony purely over WebSockets—SIP + RTP is the correct approach. • Latency: ◦ Route only what the model needs (e.g., mix or single-speaker audio) to reduce bandwidth/processing and keep latency low. • Security: ◦ Use secure SIP (TLS/SRTP) for trunking and authenticated WebSocket connections to the model API. • Features: ◦ DTMF, call transfer, recording, and hold are handled at the SIP/LiveKit level. ◦ Real‑time transcription, intent detection, or live suggestions are handled by the WebSocket model integration. • Mixing and media injection: ◦ If you need the model to speak to the caller, convert model text to TTS audio and inject that audio into the LiveKit room as a participant or as an audio track. Next steps I can help with • Walkthrough to configure a SIP trunk and LiveKit dispatch rules for inbound/outbound calling. • Example code or sequence for pulling audio from a LiveKit room and streaming it to a WebSocket model API. • Example for injecting TTS back into the LiveKit room. • Recommendations on audio mixing strategy (mix on server vs. per‑participant) depending on whether you want speaker‑separated transcription. Which next step would you like to start with: SIP trunk + LiveKit setup, streaming audio to a WebSocket model (example code), or TTS injection back into the call? , with interactive elements