https://livekit.io logo
Join Slack
Powered by
Can anyone tell if LiveKit provides direct integra...
# ask-ai
c
Can anyone tell if LiveKit provides direct integration with WhatsApp using SIP?
t
Thanks for reaching out, Hitansh. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Thank you for reaching out! Currently, LiveKit doesn’t offer direct integration with WhatsApp using SIP, but the good news is that the team is actively working on enabling direct WebRTC access for WhatsApp in the future, which will make connecting much easier. In the meantime, you can set up a custom SIP proxy or gateway to bridge WhatsApp calls into LiveKit rooms. This approach uses SIP-compliant infrastructure to translate media streams and connect through SIP trunking. If you'd like, I’d be happy to guide you through the details of setting up SIP integration with LiveKit—just let me know! Sources: lib/livekit/proto/livekit_sip_pb.rb | README.md | SIP Integration | livekit/livekit_composite | DeepWiki | Agents telephony integration | LiveKit Docs | Agents telephony integration | LiveKit Docs
c
What is the ETA for this?
If you'd like, I’d be happy to guide you through the details of setting up SIP integration with LiveKit—just let me know! - Yes please
t
We’re excited to share that LiveKit is actively working on enabling direct WebRTC access for WhatsApp! While we don’t have a specific ETA or release date just yet, please know that our development team is dedicated to making sure the integration is smooth and reliable. In the meantime, if you need to connect right away, there’s a helpful workaround using a custom SIP proxy to bridge WhatsApp calls into LiveKit rooms. I’d be happy to provide more details on how this works—just let me know! Sources: WhatsApp Integration with LiveKit
I'd be happy to guide you through setting up SIP integration with LiveKit! Here’s a simple step-by-step approach to get you started: 1. Create a SIP Trunk: Begin by using the LiveKit API or CLI to define trusted sources and set up authentication for your SIP traffic. This step is important to make sure only authorized SIP calls come through to LiveKit. 2. Set Up a SIP Dispatch Rule: Next, configure how incoming SIP calls should be directed into specific LiveKit rooms. You can easily do this using the 
CreateSIPDispatchRule
API. 3. Translate Media Streams: If you’re bridging WhatsApp calls, just make sure your SIP proxy can handle Opus-encoded audio and convert it into a format that works smoothly with LiveKit. 4. Monitor Call Status: You’ll be able to identify SIP participants in LiveKit by their identity prefix 
sip_
. Tracking their call status, like dialing or active, is straightforward through participant attributes. 5. Handle DTMF and Signaling: LiveKit supports DTMF tones and SIP REFER (call transfer), which are really handy for interactive voice systems. To make things even easier, I recommend checking out the LiveKit SIP example application. It’s a great practical example that shows a full web-based phone calling interface built with Next.js and React. If you have any questions along the way or need help tailoring this to your specific needs, please don’t hesitate to reach out—I’m here to help!
c
In the meantime, if you need to connect right away, there’s a helpful workaround using a custom SIP proxy to bridge WhatsApp calls into LiveKit rooms. I’d be happy to provide more details on how this works—just let me know! - Please do so
t
I’m happy to guide you through how to bridge WhatsApp calls into LiveKit rooms using a custom SIP proxy. Here’s a friendly step-by-step overview: 1. Set Up a SIP Proxy: First, you’ll want to deploy a SIP proxy that can connect with WhatsApp’s calling APIs or third-party tools. Think of this proxy as the helpful middleman that manages signaling and media translation. 2. Translate Media Streams: It’s important that the SIP proxy can handle Opus-encoded audio (which WhatsApp uses) and convert it into RTP streams that LiveKit understands. 3. Configure SIP Trunk: Next, use LiveKit’s SIP API to create a SIP trunk. This step sets up trusted sources and authentication for SIP traffic to keep everything secure and smooth. 4. Route Calls: Then, set up SIP dispatch rules in LiveKit so incoming calls from the SIP proxy are directed into the right LiveKit rooms. 5. Monitor and Manage: Finally, keep an eye on SIP participants in LiveKit by their identity prefix (
sip_
) and manage their state, whether they’re dialing or active. I understand this setup involves some custom infrastructure and development work, but it’s a great way to enable interoperability until WhatsApp offers direct WebRTC access. If you’d like, I’m here to help with any further guidance or resources you might need!
h
hi you can checkout this blog : https://pypeai.com/blog/whatsapp%20call
👍 1
t
Thank you so much for sharing the blog link! If you have any questions or need a hand with integrating WhatsApp calls into LiveKit, please don’t hesitate to reach out. I’m here to support you every step of the way!
5 Views
Open in Slack
PreviousNext
Powered by