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.
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 →
What is the difference between hiring a WebRTC developer and a VoIP developer?
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.
Can you scale a WebRTC application beyond a handful of participants?
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.
Our calls connect but audio or video sometimes fails. Can you fix that?
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.
Can a WebRTC developer connect our browser app to normal phone numbers?
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.
Written by the Engineers
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Read Article →Ready to Hire a WebRTC Developer?
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