Hire a WebRTC Developer
Hire WebRTC developers who build browser-based voice and video: LiveKit, mediasoup, Janus and Jitsi SFUs, TURN/STUN under real NAT, simulcast and bandwidth adaptation, recording pipelines, and bridging browser sessions to the PSTN.
What Your WebRTC Developer
Brings to the Table
Every WebRTC Developer we place has been vetted for real-world production experience — not just resume keywords.
WebRTC Developer Roles
You Can Hire
"WebRTC Developer" covers several different jobs. Tell us which problem you are solving and we will put the right one on it.
WebRTC Application Developer
Builds the browser side - getUserMedia, peer connections, track handling, reconnection when a laptop lid closes or a phone changes network. The layer your users actually touch.
Media Server Engineer
Deploys and scales the SFU: LiveKit, mediasoup or Janus. Sizes rooms, configures simulcast layers, and handles autoscaling so a spike does not degrade every call at once.
Media Path & NAT Engineer
Lives in ICE, coturn and DTLS-SRTP. The engineer for calls that connect but carry no audio, because the fault is almost always traversal rather than application code.
WebRTC-to-SIP Gateway Engineer
Bridges browser sessions to real phone numbers through Kamailio or FreeSWITCH with RTPEngine - codec transcoding, DTMF in both directions, and call control that matches on each side.
What Clients Hire
a WebRTC Developer For
Projects we have actually delivered, not a capability list. If yours is not here, it is still worth asking.
Group video calling
Peer-to-peer stops being viable past four or five participants. Routing through an SFU with simulcast keeps weaker clients from being forced to the highest bitrate.
Browser softphone for an existing PBX
A dialler in the browser that registers against Asterisk or FreeSWITCH over SIP-in-WebSocket, so agents work without a desk phone or a desktop client.
Sub-second live streaming
When HLS latency of several seconds breaks the product - auctions, trading, live coaching - WebRTC delivers in well under a second, at the cost of a more demanding media path.
Recording and egress pipelines
Server-side composite recording, per-track archives and RTMP egress to a CDN. Usually retro-fitted after launch, which is exactly when it is most painful to add.
Telehealth and regulated video
Consultation video where the recording, retention and access rules matter as much as the call quality, and the media path has to be auditable.
Calls that connect but have no audio
One-way or no audio on a fraction of calls. We debug from real packet captures rather than guesswork - a misconfigured TURN server and dropped UDP are the usual culprits.
AI voice agents inside browser calls
Putting a real-time agent into a WebRTC session: audio forked to STT, LLM response, TTS returned into the same room within the latency budget a conversation tolerates.
How We Work Together
Dedicated Developer
One or more developers work exclusively on your project. Full-time, your timezone, your tools, your standups.
Best for: Ongoing product developmentSprint-Based
Structured 2-week sprints with defined deliverables, daily updates and fortnightly demos. Perfect for defined projects.
Best for: Fixed-scope projectsConsulting & Audit
Expert review of your existing system — architecture, code quality, performance or VoIP configuration — with a remediation plan.
Best for: System reviews & troubleshootingYour Developer Ships Faster
with AI Coding Tools
Every developer we place is trained and actively uses the latest AI coding assistants — delivering 30–50% faster output without sacrificing quality.
Common Questions
About Hiring
Still have questions? Chat with our team →
A VoIP developer works in SIP and PBX territory — Asterisk, FreeSWITCH, Kamailio, trunks, dial plans. A WebRTC developer works in the browser and the media path — SFUs, ICE and TURN traversal, codec and simulcast decisions, and getting acceptable quality on real networks. The two overlap when a browser session has to reach the PSTN, and several of our engineers cover both.
Yes. Peer-to-peer stops being viable at around four or five participants, so calls are routed through an SFU — LiveKit, mediasoup or Janus depending on what you already run. Our developers handle SFU deployment and autoscaling, simulcast layers so weaker clients are not forced to the highest bitrate, and the recording and egress pipelines that usually get added later.
That is almost always ICE and NAT traversal rather than the application. It usually means a misconfigured or overloaded TURN server, a firewall dropping UDP, or an SDP negotiation problem. We debug from real packet captures rather than guesswork, which is the same approach behind our published guide to one-way audio in Asterisk.
Yes — that bridge is a large part of what we do. A browser session terminates on a media server, then crosses to SIP for the PSTN leg, typically through Kamailio or FreeSWITCH with RTPEngine handling the media. It means codec transcoding, DTMF passing correctly in both directions, and call control that behaves the same on both sides.
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Read Article →Ready to Hire a WebRTC Developer?
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