Low latency live streaming

Low-latency streaming. More live interaction.

For an auction, a class or an interactive event, what matters is how quickly viewers see what happens. XtreamCast provides managed streaming servers with RTMP/SRT ingest and optional WebRTC playback. Choose the right workflow for your audience and test end-to-end latency before going live.

Choose the protocol for the receiving endpoint

SRT connects production equipment; HLS and WebRTC deliver video to browsers. Reducing ingest latency alone does not remove the player buffer.

WorkflowWhen to choose itWhat to check
WebRTCAuctions, classes and browser experiences that need immediate interaction.Subsecond target under suitable conditions; measure with your devices and networks. Per-channel add-on.
HLSTV channels, events and distributed audiences that prioritize compatibility.Playback uses a buffer; measure delay, stability and data transfer.
LL-HLSHTTP delivery when you need to reduce the HLS buffer.Availability depends on the origin and channel; confirm enablement with support.
SRTEncoder ingest and professional links between production systems.Packet loss, RTT, recovery buffer and authentication. Not native browser playback.

In the current configuration, WebRTC cannot be combined with LL-HLS or ABR transcoding on the same channel. Standard HLS can remain available. For multiple quality levels or large audiences, review the workflow with support.

Prepare your broadcast in four steps

  1. Define the interaction. Determine acceptable delay and expected viewers. A live conversation and a TV channel have different requirements.
  2. Connect your encoder. Send video over RTMP or SRT using your channel credentials. Match resolution, bitrate and keyframes to your actual connection capacity.
  3. Configure playback. Use the HLS player or enable the WebRTC add-on. Test the link and embed on the devices your audience will use.
  4. Measure before the event. Test with viewers in your target regions, record the results and allow time to adjust your settings.

Measure streaming delay, not just ping

A reproducible test covers the entire path: camera, encoder, server, network and player. A server ping does not measure that path.

  • Film a stopwatch and compare the source with playback in the same recording; avoid comparing unsynchronized device clocks.
  • Record the date, region, ISP, connection, encoder, resolution, bitrate, protocol and viewer device.
  • Repeat the test for several minutes and on different networks. Record typical delay, variation, startup and reconnects.
  • Test chat, bidding or user actions separately: video streaming does not automatically synchronize those applications.

What your plan covers and what your project needs

  • Video plans include RTMP/SRT ingest, HLS delivery, a player and the transfer and storage allowances listed on the pricing page.
  • WebRTC is a per-channel add-on. Check current pricing and compatible options at checkout.
  • International delivery supports viewers in different countries; distance and connectivity still affect results.
  • Bidding systems, chat, paid access and user authentication are separate integrations to evaluate.

No internet video workflow has zero latency. The practical goal is to reduce delay to a level that suits the interaction. WebRTC can achieve subsecond latency under suitable conditions; test your encoder, network and viewer setup.

SRT can improve contribution from the encoder to the server. If playback uses HLS, the HLS buffer also contributes to delay. Consider WebRTC output for browser interaction.

No. Latency depends on the complete workflow, network, device and location. An indicative figure is not a worldwide guarantee. Test the regions your event needs before choosing a configuration.

Create a three-day TestDrive and ask support for a WebRTC demonstration. Confirm that it can be enabled and is compatible with your trial channel.

Have more questions?

Our team is available 24/7 to help you with any question about your streaming project.

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