META | Link

WhatsApp Cloud API Calling Now Available

Having launched on July 1st, the new WhatsApp Business Calling API enables businesses to integrate WhatsApp VoIP calls into their applications. This release marks WhatsApp's entry into programmable voice communications, giving developers access to WhatsApp's massive user base for voice interactions. The API supports both inbound and outbound calling patterns, opening new possibilities for integration.

WISH WORKING GROUP GITHUB | Link

WHEP Protocol Debate: Server vs Client-Initiated SDP Exchange

This pull request is now a little old, but recently there has been some new activity going on. It proposes adding both server-initiated and client-initiated SDP exchange modes to the WHEP specification. The discussion has a heated debate about whether the flexibility of this approach will discourage developers from fully implementing the protocol. The thread is worth tracking for anyone actively working with WHEP.

MEETECHO BLOG | Link

Node.js WHIP/WHEP server libraries

Lorenzo Miniero has refactored his WHIP and WHEP servers from standalone Node.js applications into reusable libraries, making them available on npm as janus-whip-server and janus-whep-server. These libraries allow developers to easily integrate WHIP/WHEP functionality into existing applications, with the ability to use both protocols in the same application for complete WebRTC broadcasting solutions.

ATTILA BLÉNESI ON X | Link

Grok Voice Mode Made With WebRTC and LiveKit

xAI implemented a voice mode in Grok for Android users. According to the xAI engineer, it uses WebRTC infrastructure powered by LiveKit under the hood. The implementation probably leverages LiveKit's framework for realtime voice agents, using WebRTC for transport.

The moment we've been waiting for: full RTC.ON lineup is up on 🔥

From a Superbowl streaming case study through observability in WebRTC, a hacker's view on your media relay and many more – I must say, this year's lineup is better than it ever was before.

See full RTC.ON 2025 lineup.

And since all speakers are now announced, we're officially increasing the prices on Sept 18 – this means it's your last chance to grab the Early Bird ticket.
Use code NEWS20 to get an additional 20% off for the last two weeks of Early Birds – the price won't get lower than that!

🚀 WHAT: RTC.ON Conf 2025

🕕 WHEN: 17-19 September 2025

📍 WHERE: Kraków, Poland

Today we want to share one of the multimedia projects we've worked on – helping Turn.io bring real-time audio to their WhatsApp-based messaging platform.

Turn.io is a Public Benefit Corporation that enables NGOs and governments to reach underserved communities through WhatsApp. When they approached us, they already had a powerful system built on Phoenix, LiveView, and React, handling high-scale text communication. What they needed next was a way to support real-time voice calls – seamlessly, reliably, and within their existing architecture.

We joined forces to make it happen. Over the course of the project, we extended the platform to support browser-to-Turn.io-to-WhatsApp audio routing using Elixir WebRTC, enabling voice calls between web agents and mobile WhatsApp users. We didn’t stop there. We also added:

Live transcription using OpenAI, streaming audio via RTP and returning transcripts in real time.

A smart AI assistant that could handle conversations before handing over to human agents – all without dropping the connection.

DTMF tone recognition, allowing users to press numbers on their keypad during calls (especially useful for IDs and reference numbers).

Post-call analytics using get_stats() stored in ETF format, covering bytes sent, packet loss, ICE candidate data, and more.

To ensure this worked across all networks – including those with restrictive NAT types – we added support for Cloudflare TURN servers and improved Elixir WebRTC with DTLS certificate fragmentation (fixing MTU issues with providers like Orange).

One tricky challenge was integrating Elixir WebRTC with OpenAI’s Go-based Pion implementation. They didn’t talk to each other out of the box – so we patched Elixir WebRTC to support aggressive ICE nomination, reducing connection setup time dramatically.

The results? Weekly voice call volume grew by 250%, and Turn.io’s support team gained powerful debugging tools via LiveDebugger and post-call stats. More importantly, they now offer a smooth, AI-enhanced calling experience to users in healthcare, education, and crisis response sectors – directly on WhatsApp.

The project was a joy to work on – deep Elixir tech, real-world reliability challenges, and a mission-driven client.

Fancy some help with your stack? We’re available – and excited to tackle your toughest multimedia-related (and not only) problems.

Although we work with a pretty wide tech stack when it comes to streaming and multimedia, Elixir has always been one of the technologies we’ve absolutely loved. So, we decided to combine the best of both worlds and create an event that brings these two together – and that’s how Global Elixir Meetups were born.

The idea originally came from a group of Elixir contributors. As soon as we heard it, we immediately thought of Fishjam – the low-latency live streaming and video conferencing API we’ve built– and knew it would be the perfect fit. So, we rolled up our sleeves, gathered a team, and got to work!

GEMs were created to reignite the Elixir community around the world. Even if you can’t attend any of the meetups in person, you can still participate by joining one of the virtual streams.

Check out our meetup map and see how GEMs are lighting up the globe:

We hope you enjoy the RTC.ON newsletter as much as we do. It's great to see all of you joining us each month for a little multimedia walk through – thanks!

Here are some more ways to connect with us:

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