HD Group Audio Chat

The WebSocket version turns any conversation, group chat, or chat room into a multi-party voice call supporting up to 50 simultaneous participants. The feature uses WebRTC with a dedicated media server for efficient routing — call quality stays high even with dozens on the line, while bandwidth stays modest compared to video.

WebSocket version

Group audio chat is a WebSocket-version feature. The free version does not include voice calling.

What it adds#

Up to 50 simultaneous voice participants per call
Works in private one-on-one conversations, group chats, and chat rooms
Lower bandwidth than group video — typically 50–80 kbps per participant
Media-server routing (SFU model) for stable quality at scale
Remote mute/unmute control for the call host
Per-role restrictions: control which user roles can place or join calls
Optional mini-chat call buttons in members directories and on profile pages
Bandwidth-friendly fallback for users on cellular / low-quality connections

How it works#

A user clicks the Voice Call button in a conversation. Better Messages signals the request through the WebSocket relay; the cloud server allocates a call room on the media server. All participants connect to the media server via WebRTC. Each participant uploads their own microphone stream once; the media server distributes streams to everyone else — this is dramatically more efficient than full mesh (where each participant uploads to every other participant individually).

Topology	Network upload per participant
Full mesh (peer-to-peer)	(N−1) × stream — quadratic, breaks past 5–6 participants
Media server (SFU)	1 × stream — flat, scales to 50+

The host (the user who started the call) can remotely mute disruptive participants. Participants can also self-mute, self-unmute, and leave anytime.

The call session is independent of the underlying conversation thread — when the call ends, the conversation remains exactly as it was, with a system message noting "Voice call ended" and duration.

When to use group voice over group video#

Choose group voice when…	Choose group video when…
10+ participants	Visual presence matters
Mobile/cellular users	High-bandwidth connections
Discussion / audio-first content	Demo, screen-share, face-to-face
Bandwidth-constrained users	Standard broadband
Privacy preference for audio only	Trust-building, sales / coaching

Requirements#

Site must be served over HTTPS (browser WebRTC requirement)
Each participant needs a working microphone
Modern browser with WebRTC support (Chrome, Firefox, Edge, Safari, mobile equivalents)
WebSocket-version license active

Configuration#

Per-role restrictions and optional mini-chat call buttons apply to both audio and video group calls — see Group Video Chat → Per-role restrictions for the shared admin controls.

Frequently asked questions#

What is the maximum number of participants?#

50 simultaneous voice participants. Sites that regularly need more can contact support to discuss the self-hosted plan, where the cap can be raised based on media-server capacity.

Are group voice calls recorded?#

No — calls are real-time only and not recorded by default. The plugin doesn't include built-in recording. Sites that need recording typically use a screen-recording layer at the participant level or browser-based recording extensions, subject to participant consent.

Can guests (non-logged-in users) join a group voice call?#

Yes — if guest chat is enabled on the chat room, guests can join voice calls placed in that room. They appear by their entered display name.

What's the bandwidth requirement per participant?#

About 50–80 kbps upload + 50–80 kbps download per active speaker. A 20-person call uses about 1.5 Mbps download on each device. The codec is Opus.

Does the media server store anything?#

No. The media server routes audio streams in memory only — no storage, no recording, no transcription on the audio path. Call metadata (start time, participants, duration) is logged for the host's call history.