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HD Audio Calls

The WebSocket version enables HD 1-on-1 audio calls directly from any conversation. Built on WebRTC for peer-to-peer audio streaming when possible (with TURN relay fallback for restrictive networks), audio quality is high and latency is low — comparable to dedicated voice apps like Zoom or Discord.

WebSocket version

Audio calls are a WebSocket-version feature. For multi-party voice, see group audio chat (up to 50 participants).

What it adds#

  • 1-on-1 HD audio calls between any two participants
  • WebRTC peer-to-peer streaming — the audio doesn't transit Better Messages servers
  • Low latency, Opus codec for high quality at modest bandwidth (~30 kbps each direction)
  • Initiate from any conversation with one click
  • Works in all modern browsers (Chrome, Firefox, Edge, Safari, mobile equivalents)
  • Built-in call UI with mute, hang up, and timer

How it works#

When a user clicks the Call button:

  1. Signaling: the WebSocket relay carries SDP and ICE candidates between the two browsers
  2. Peer-to-peer connection: the two browsers establish a direct WebRTC connection
  3. Audio streaming: voice flows P2P between the participants
  4. Fallback: if direct P2P fails (corporate firewall, symmetric NAT), a TURN relay forwards the encrypted audio
Network scenarioWhat happens
Normal home/office networkDirect peer-to-peer — minimal latency
One participant behind strict firewallTURN relay forwards encrypted audio
Both behind corporate proxiesTURN relay (slight added latency)

All audio is DTLS-SRTP encrypted end-to-end — even when relayed through TURN, the audio bytes are encrypted between the participants.

When 1-on-1 audio calls fit#

Use casePattern
Coaching sessionCoach-client 1-on-1 call inside the existing chat thread
Customer support escalationVoice call when chat alone is taking too long
Marketplace negotiationQuick voice call to close a higher-value deal
Telemedicine consultationPractitioner-patient 1-on-1
Dating communityAudio call as a step before video

Requirements#

  • Website must use HTTPS (WebRTC requires secure context for microphone)
  • Each participant needs a working microphone
  • Modern browser with WebRTC support
  • WebSocket-version license

Frequently asked questions#

Are calls recorded?#

No — calls are real-time and not recorded by Better Messages. To record, participants can use their own screen-recording or call-recording tools (with consent). The plugin doesn't include built-in recording.

Can I configure per-role call permissions?#

Yes — restrict who can place calls via WP Admin → Better Messages → Settings → Calls. Useful for paid-tier-only calls.

What if the recipient is offline?#

The caller sees a "user offline" notice. Better Messages doesn't queue ringing for offline users.

Does audio quality degrade with poor networks?#

WebRTC adapts to bandwidth — under poor conditions, the bitrate drops automatically. Even on cellular, conversations remain intelligible. For truly bad connections, the call may drop and need to be re-initiated.

How do group audio calls differ?#

Group audio chat supports up to 50 participants and routes through a media server (SFU) rather than peer-to-peer. Different scaling model, similar end-user experience.

See also#