Voice makes timing part of correctness
In a live conversation, perfectly delivered audio can still be unhelpful if it arrives too late. Choosing transport means deciding how a system handles loss, jitter, congestion, and playback deadlines. Both WebRTC and WebSockets can carry audio; they provide different starting points for those responsibilities.
Evaluate the user-to-agent media path separately from a controlled connection between servers. A design that works well inside a data center may behave differently over mobile networks. The appropriate choice depends on the endpoints and the conditions you need to support.
What a WebSocket connection provides
The WebSocket protocol defined in RFC 6455 frames bidirectional messages over TCP. [1] TCP provides an ordered, reliable byte stream while the connection is functioning. Loss recovery can delay delivery of later bytes, creating head-of-line blocking. It does not mean every conversation will stall or that a disconnected stream is guaranteed to arrive.
WebSockets do not define audio codecs, jitter buffering, or playout deadlines. The surrounding application must provide suitable handling. Browser microphone processing can still be available to a WebSocket application: echo cancellation is not exclusive to the transport selected for sending the captured audio.
What WebRTC adds for media
WebRTC combines media negotiation, RTP transport, connectivity establishment, and media security requirements. [2] Implementations provide media-oriented timing and buffering behavior. The suite is designed for interactive audio and video, where recovering very late data may be less useful than keeping playback current.
UDP is an important transport path, but WebRTC can also use relay and TCP paths when network restrictions require them. [3] Those fallback paths still need measurement. The label WebRTC does not remove the effects of packet loss, buffering, physical distance, or shared network capacity.
Understand relays and routing separately
A selective forwarding unit routes media between participants and can be useful for multi-party sessions or centralized processing. It is not a prerequisite for a two-party WebRTC connection, nor does it make media routing free. Its placement, traffic volume, and failure behavior affect the system.
A nearby relay can improve connectivity or routing in some cases. It cannot eliminate the time needed to reach a distant model or application service. Follow the entire route from microphone through inference to speaker when deciding where components should run.
Choose using failure conditions
Compare connection success, playback delay, jitter-buffer behavior, interruption response, and recovery under changing bandwidth. Include packet loss, competing traffic, and networks that block UDP. Standards for evaluating interactive congestion control provide useful test dimensions [5] rather than a promise that one stack always wins.
WebRTC is a strong starting point when browser or mobile media behavior is central to the application. WebSockets can remain appropriate for signaling, structured events, controlled server links, and deliberately engineered audio paths. Select the transport with the evidence and operational responsibilities in view, then test the complete conversation rather than the socket alone.
Research and references
- [1] RFC 6455: The WebSocket Protocol ↗
- [2] RFC 8834: Media Transport and Use of RTP in WebRTC ↗
- [3] RFC 8835: Transports for WebRTC ↗
- [4] RFC 8836: Congestion Control Requirements for Interactive Real-Time Media ↗
- [5] RFC 8868: Evaluating Congestion Control for Interactive Real-Time Media ↗
Technical review: November 9, 2026. Research supports the underlying concepts; implementation recommendations and illustrative scenarios are editorial guidance, not measured product results.