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Local Live & Playback with WebRTC
Local Live & Playback with WebRTC

Local live & playback reduces bandwidth usage while providing web portal users a lower latency experience

Updated over a week ago

Introduction

WebRTC (Web Real-Time Communication) is a technology that enables direct communication between applications by facilitating peer-to-peer connections, allowing for smooth transmission of video and audio data over the internet. One of its notable features is the ability to facilitate local live viewing and playback.

Local Live & Playback

WebRTC enables real-time viewing of local live video feeds directly from your browser, ensuring minimal delays by accessing the video feeds directly from the gateway or camera-to-cloud (C2C) device. It also provides quick access to recorded video content with low latency, allowing you to review past events with reduced buffering.

Cloud gateways can use local playback for buffered and locally stored data on the device. The duration of local video retention depends on estimated maximum retention time or until the disk is full. If playback is requested outside the local data availability period, cloud playback is activated.

When WebRTC is active, it will locate and utilize the most efficient paths for media routing, reducing bandwidth consumption. If a user is viewing video streams from a device on the same network as the gateway or C2C device, the web app switches to Local Live and Local Playback modes. Keep an eye on the banner at the top right of your video player – it indicates local viewing mode. However, if a user is accessing a camera’s footage remotely and local live streaming and playback is not feasible, the system will seamlessly transition to cloud-based live streaming and playback.

Requirements for Local Live & Playback

  • There must be a direct connection between the client and the camera without any intermediaries or proxies in between.

  • Your gateway/device has an internet connection for credential exchange.

  • Your network will need to allow UDP traffic to and from the gateway to the local client on the following ports: 20000 - 24999

  • Both the clients and gateways will need access to UDP traffic to *.arcules.com over port 443. For more details visit Network Readiness Guide.

  • Additionally, WebRTC must be enabled at either the organization or user level.

How to enable WebRTC

Users assigned the IT Manager role possess centralized control over the organization's streaming protocol preferences within the Organization Settings page, accessible under Settings. These settings apply universally to all users within the organization.

Users within the organization have the flexibility to tailor their WebRTC settings according to their network conditions. If the organization-wide WebRTC setting presents difficulties due to specific network configurations, users can override this setting by accessing their user profile icon. This ensures a smooth and optimal video streaming experience tailored to their individual needs.

Note: Disabling WebRTC will impact live audio functionalities.

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