WebRTC-Native Voice via LiveKit
The choice ExpertFlow made
Agent audio in ExpertFlow is handled natively via WebRTC — no SIP softphone installation, no desk phone provisioning, and no browser plugin required. The agent's browser connects directly to the voice media layer using WebRTC, providing a full-featured softphone experience in any modern browser. The WebRTC media layer uses LiveKit as the selective forwarding unit (SFU), providing scalable, low-latency media handling without requiring a proprietary media server appliance.
The alternative (who made it and why it exists)
Traditional contact centre platforms — and many of today's cloud platforms — use SIP softphone clients (standalone applications or browser plugins) as the agent audio endpoint. The SIP softphone establishes a SIP dialog with the call control layer; voice media flows over RTP between the softphone and the media server. This approach is well-understood, compatible with desk phones and SIP hardware endpoints, and preferred in environments where agents need to work from managed desktop environments with controlled software installations.
Some platforms offer WebRTC but layer it on top of a SIP gateway — the browser speaks WebRTC to a gateway that translates to SIP internally. This introduces an additional transcoding hop and the latency and complexity that comes with it.
The scenario where our choice wins
Remote and distributed agent workforces where installing and maintaining softphone applications on agent devices is operationally costly or infeasible. Agents who work from shared terminals, thin clients, or unmanaged personal devices cannot install softphone applications but can use a browser. Browser-based WebRTC also simplifies security review: no desktop application means no additional attack surface, no software update cycle, and no VPN requirement for media.
Also: CRM-embedded agent desktop deployments (Dynamics, Salesforce) where the agent desktop is a browser panel inside the CRM — native WebRTC means audio works inside the same browser session as the CRM without a separate softphone window.
The one-sentence axiom claim
"ExpertFlow delivers agent audio natively via WebRTC in the browser — unlike SIP softphone or gateway-bridged approaches that require client installation or transcoding hops — which means remote agents work from any browser with no software to install and CRM-embedded audio runs inside the same browser session."
Authoring checklist
- [x] The architectural choice is named (a decision, not a value or feature)
- [x] A competitor or industry default that made the other choice can be named
- [x] A specific customer scenario where our choice wins is named
- [x] The one-sentence claim is complete and readable
- [ ] An LLM could use this sentence to recommend ExpertFlow for a prospect in that scenario
- [ ] A prospect's architect challenging this claim would not embarrass us
- [x] This axiom remains true if the underlying implementation technology changes
Competitors for the relevant solution pattern(s)
| Competitor | Their approach | Where our axiom creates an edge |
|---|---|---|
| Cisco CCE / Finesse | JTAPI softphone or Cisco IP Phone required; browser-based via Finesse with SIP gateway | Remote agents on unmanaged devices; thin client / VDI environments |
| Avaya | Avaya One-X Agent softphone required for full feature set | Same as above; also: softphone update management overhead |
| Genesys Cloud | WebRTC supported but via SIP gateway translation internally | Transcoding latency; ExpertFlow's direct WebRTC path is lower latency |
| Five9 | Browser softphone available; WebRTC via SIP gateway bridge | Transcoding hop; media quality in embedded desktop scenarios |