Years of WebRTC expertise, complex dev-ops, and product development
in simple "low code" snippets that you can embed in any website or JS app.
The only WebRTC video product for developers that hits the sweet spot between "low-code," and "highly-custom."
Ever thought: "What if I could embed Zoom into my app and then build new products around it?" Then you tried Zoom's developer tools and it didn't go as planned? Call off the search. You just found SignalWire.
Our unique "server side mixing" unifies up to 300 video and audio streams in the cloud and distributes a single stream back to every participant. The upshot? 50-80% less bandwidth and CPU consumption on the client-side + <50ms round-trip latency = none of the laggy video and audio you see with Zoom, Meet, and MS Teams.
Instead of 20 active participants and a few hundred passive spectators, support 300 active video and audio streams in every room...and run hundreds of rooms at the same time.
3-4 prebuilt video layouts may work for basic video calls, simple webinars, and small virtual classrooms. But when your production values need that "premium" vibe, you'll want more granular options. We offer 12 today, on the way to 20.
Sub-50ms round trip latency in your video apps means way more milliseconds available for real time natural language and machine vision models.
Unified Stream Video Conferencing mixes video and audio in the cloud and delivers a shared experience to every participant, consuming a fraction of the bandwidth and CPU of the standard approach.
Traverse firewalls. Maintain <50ms round trip latency while minimizing bandwidth and CPU consumption on your customers' devices. Auto adjust bitrates to maximize quality in unpredictable networking conditions. Do it all without worrying about any of that yourself.
Embed them once and watch them sprout powerful new features over time. Little to no new dev work on your part required.
The Prebuilt Video API comes with optional built-in chat, offering basic 1:1 and group chat features to video conferences without adding another API to the stack.
Native SIP and PSTN integration means your video conferencing rooms can extend an existing telecom stack without changing vendors or adding new equipment, conference in voice callers via SIP, dial out to old-school phone numbers, and send SMS invites with links to join the room.
100% of features available with no upfront fees or subscriptions required.
Rates are per-minute, per-participant.
|MONTHLY MINUTES||STANDARD (720p)||FULL HD (1080p)|
|1M or more||$0.00510||$0.00610|
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