the agent seems to lag or have a bit laggy voice on SIP phone calls using twilio

Thanks for reaching out, Vanxh. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.

how do i fix the config for better latency and quality

To help you get the best latency and call quality for your SIP phone calls using Twilio in LiveKit, here’s a friendly step-by-step guide: 1. Boost Call Quality: Try enabling HD Voice in LiveKit’s SIP settings—this works great if your SIP trunk and endpoint support it. 2. Keep Things Close: Make sure your LiveKit server, Twilio, and any other services are all hosted in the same region. This really helps cut down on latency. 3. Set Up Your Twilio SIP Trunk: ◦ Create a SIP trunk with a domain name that ends in

<http://pstn.twilio.com|pstn.twilio.com>

. ◦ Point the origination URL to LiveKit’s SIP endpoint. ◦ Link your Twilio phone number to this SIP trunk. 1. Configure Inbound Trunk and Dispatch Rule in LiveKit: ◦ Create an inbound trunk and add your authentication details. ◦ Set up a dispatch rule to send incoming calls to the right rooms or agents. 1. Keep an Eye on Latency: ◦ Watch important metrics like Time To First Token (TTFT) and Time To First Byte (TTFB). ◦ Use automatic video quality management to handle any network hiccups smoothly. 1. Double-Check Your Environment Setup: Make sure all your environment variables—like Twilio API keys and SIP credentials—are correctly added in your

.env.local

file. 2. Test It Out and Tweak: Use the LiveKit SIP agent example to test your setup and see how the call quality feels. If you want me to walk you through any of these steps or need help with specific settings, just give me a shout—I’m here to help! Sources: Create and configure a Twilio SIP trunk | LiveKit Docs | .env.example | AI Agents Framework | livekit/livekit_composite | DeepWiki | Voice Assistants | livekit/livekit_composite | DeepWiki

how can ii configure livekit sip location

Hey there! To get your LiveKit SIP location all set up, just follow these simple steps: 1. *Create SIP Trunk and Dispatch Rule:*`# Create SIP Trunk livekit-cli create-sip-trunk \ --request <path to sip trunk creation request json file> # Create SIP Dispatch Rule livekit-cli create-sip-dispatch-rule \ --request <path to sip dispatch rule creation request json file>` ◦ Start by using the LiveKit CLI to create a SIP trunk and dispatch rule: 1. *Configure SIP Service:*`api_key: <your-livekit-api-key> api_secret: <your-livekit-api-secret> ws_url: <your-livekit-ws-url> redis: address: <redis-address> username: <redis-username> password: <redis-password> db: <redis-db>` ◦ Next, make a YAML configuration file with these fields: 1. *Set Environment Variables:*`LIVEKIT_API_KEY LIVEKIT_API_SECRET LIVEKIT_WS_URL LIVEKIT_SIP_TRUNK_ID (if using SIP trunking)` ◦ Don't forget to set these environment variables: 1. *Create Dispatch Rule:*`{ "rule": { "dispatchRuleDirect": { "roomName": "my-new-room" } }, "trunkIds": ["<trunk-id>"], "hide_phone_number": true }` ◦ This is where you define how incoming calls get routed to your LiveKit rooms. Here's an example: 1. Verify Configuration: ◦ Make sure your LiveKit SIP URI is correctly set up in your SIP provider settings. ◦ Double-check that your SIP provider is sending calls to the right LiveKit SIP endpoint. If you want to dive deeper, check out the configuration guide. And hey, if you run into any questions or need a hand, just give me a shout—I’m here to help!

also can you explain if i am getting dealyed responses from the voice pipeline

Hey there! If you’re noticing some delayed responses from the voice pipeline in LiveKit, no worries — here are a few friendly tips to help smooth things out: 1. Handle RAG Delays: ◦ Try using system prompts to gently let users know when the agent is busy looking something up. ◦ Keep users engaged by showing “thinking” messages, whether static or dynamic, while processing is happening. ◦ Playing short audio clips can be a nice non-verbal way to signal that something’s happening behind the scenes. 1. Optimize Latency: ◦ Cut down on delays in Speech-to-Text (STT), GPT response generation, and Text-to-Speech (TTS) by choosing efficient models and settings. ◦ You can tweak stream delays with the

ApmSetStreamDelayRequest

protocol to get smoother audio streaming. ◦ Adding interrupt handling helps stop any unresponsive TTS playback, making things feel more responsive. 1. Turn Detection: ◦ Using turn detection and endpointing logic can really help reduce wait times in voice chats. LiveKit has built-in support for VAD, STT signals, and even custom models to nail accurate turn detection. If you want to see all these strategies in action, check out the RAG Voice Assistant Demo. And hey, if you have any questions or want help tweaking your setup, just give me a shout — I’m here to help! Sources: packages/livekit-rtc/src/proto/audio_frame_pb.ts | agents/src/pipeline/pipeline_agent.ts | Voice Assistants | livekit/livekit_composite | DeepWiki | delay-handling/agent.py | Realtime model integrations | LiveKit Docs