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Advanced communications from the source. | SignalWire

Don't let conventional WebRTC assumptions limit your possibilities

Most approaches to WebRTC video conferencing use a limiting approach called SFU.

SFU Muddies Up The UX

SFU devours bandwidth, device CPU, and battery. SFU delivers individual video and audio streams of each participant to every other participant, leaving the work of decoding and rendering the media to the device CPUs. So with 5 participants, every participant receives and decodes 4 video streams. 16 participants? 15 streams. You see where this is going?

This MCU makes YOU the superhero

The SignalWire Video Conferencing API uses an MCU to mix all video and audio in the cloud and distribute a single stream back to every participant. 50-80% less bandwidth and compute consumption on the client-side + <50ms round-trip latency + none of the video and audio artifacts seen with SFU = unified, high production value experiences for large audiences.

Deliver Premium Production Values To Large Audiences

300 people per room

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.

12 pre-built video tile layouts

3-4 prebuilt video layouts may work for basic video calls, simple webinars, and small virtual classrooms. When your audience is large and your production values need to be premium, you'll want more granular options. We offer 12.

Modify the stream on the way out

Run your video apps in real-time through natural language and machine vision models and deliver utterly original experiences

    Choose SignalWire when your team, brand, and audience won't settle for janky.

    Realize your most inspiring visions

    Our unique cloud multiplexing empowers you to integrate video conferencing rooms for up to 300 people per room (and run hundreds of those at once)...adding audio effects and XR overlays or running real-time NLP and machine vision models...all without consuming your customers' bandwidth and CPU.

    Let us handle the complexity of WebRTC

    Build video conferencing features that scale, traverse firewalls, work across browsers, smartphones, and (eventually) XR devices. Integrate with your existing apps and phone systems. Maintain <50ms latency while minimizing bandwidth and CPU consumption on your customers' devices. Do it all without worrying about any of that yourself.

    Native chat, too

    Coming Soon

    With the Chat API, add basic 1:1 and group chat features to video rooms without adding another API to the stack.

    • 1:1 and group chat
    • Send messages, emojis, and files
    • Automatic 7-day message retention
    • Server side access controls for users and devices
    • Real-time events when messages are received

    Integrate with existing phone systems and call centers

    Coming Soon

    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.

    Start with an SDK or a single snippet of code

    Customize everything precisely as you wish with our SDKs or ship a basic video conferencing feature with Embeddable Rooms.

    Native SDKs for precision customization

    Precisely craft every aspect of your video conferencing features with our Javascript, Node, and React SDKs

    Or get started with a snippet of code

    Embed video conference rooms for moderators and guests into any webpage in minutes

    Pay-as-you-grow pricing

    100% of features available with no upfront fees or subscriptions required.

    Video Conferencing API Pricing

    Rates are per-minute, per-participant.

    STANDARD (720p)FULL HD (1080p)

    Start building

    Contact us to get started.

    Talk to an expert