Low Latency

Live Streaming

Infrastructure for live streaming.

Scale to 100+ million viewers and beyond

  • €0

    Adaptive bitrate encoding at no cost

  • 0,3-4 sec latency

    Scalable and reliable streaming


    Simple dashboard, SDK and API

Workflow ready

Scale with us, if you have your own video server and apps. And use our infrastructure with open source demos, if you need to build a new video service from scratch.


RTMP, SRT, WebRTC or multicast broadcasting of your streams to million of viewers.

Scroll horizontally to see the diagram

Simple steps to start

Step 1
Create a stream

Get a stream key for PUSH or PULL in the Dashboard

or use API to get a secret

POST https://api.gcore.com/streaming/streams
Request: { "stream": { "auto_record": true,"active": true} } Response: { "id": 309827,"push_url": "srt://live.gcore.com/secret","backup_push_url": "rtmp://live2.gcore.com/secret" }
Step 2
Send a stream

Gcore receives RTMP, SRT, HLS, WebRTC, UDP Multicast, and other protocols, which are supported by most broadcast software/hardware as well as open source software for mobile applications.

  • Web studio in just browser
  • Streaming from mobile apps (open source demo & howto)
  • OBS and vMix parameters
Step 3
Watch a stream

We prepare adaptive bitrate for devices and bandwidth automatically. Use our HTML-player for free or insert manifest into your player to play LL HLS, MPEG-DASH CMAF streams.

src="https://play.gcore.com/live/player" allow="autoplay; encrypted-media" allowfullscreen />

Ready to get started?

14 days free trial

Fast delivery

You get cutting-edge streaming technology without investing in expensive infrastructure. Delay in delivery to the end viewer up to 4-5 seconds. At the same time, high quality audio and video up to 4K/8K is maintained.


For video content that is very sensitive to delays (sports, gaming, news, auctions, interactive shows), we use top-notch technology to minimize such disruptions: LL HLS, MPEG-DASH CMAF, HESP and low-latency CDN


Video Capturing
  • SRT, RTMP input
  • MPEG2 UDP multicast input
  • WebRTC input from browser
  • PUSH & PULL input
  • Backup input of live streams
  • Adaptive mobile RTMP
  • Video conferencing input
  • Realtime encoding
  • Adaptive bitrate
  • Recording
  • Video hosting
  • Low Latency transcoding
  • Video 360
  • Server-Side Video Insertion (SSAI)
  • Statistics and analytics
  • Pure CDN delivery
  • Low Latency Streaming
  • Restreaming to social networks
  • Video protection
  • Playback on iOS, Android, Windows, Mac, Linux, Set Top Box, Smart TV, and game consoles
  • HLS, MPEG-DASH, fragmented MP4
  • DVR
  • HTML5 player
  • Monetization (CSAI, VAST/VPAID)
  • Anti Adblock

Frequently Asked Questions

You need a streaming software/videocam that sends streams in rtmp/srt.
Or if you’re a pro - you may have a pre-set up server from which we can extract your srt/rtmp stream.

After we get your stream, we transcode it in the qualities you need and to the http-compared protocol (HLS/DASH) which are suitable for web-players and CDN.
Then we deliver this stream to your audience anywhere in the world using our CDN.
We provide you with an iframe of our player to embed into your application. But that’s not mandatory. You can use any other hls.js/dash.js compatible player to place our playlist in.

Here is a step-by-step guide explaining what settings you need to set up in the Control Panel .

We accept any kind of software or videocams. It doesn’t matter what you use - free OBS or Entreprise LiveU, or just a videocam with a pre-set up software.

you just need to make sure that your software works with rtmp or srt. Most modern softwares surely accept these protocols.

If you have something ve-e-ery specific just contact us.

Sure, you can. This protocol is much more stable than RTMP - it doesn’t drop connection if there’s smth wrong with provider routes.
We recommend all our customers to use SRT. But rtmp also works as the most common streaming protocol.
Check, please, whether your software supports SRT. Then send a stream to us in that protocol. For your and our serenity.

From 1 to 100 000 000 or even more. There’s no end-point.
We use our own EdgeNetwork. With PoPs all over the world united into redundant clusters. So our edge-servers share load between each other. That means, that viewers are separated between lots and lots of servers, based on:

  • Geography/Topography – which means users get the stream from the nearest location (both geographical and providers routes)
  • Overload - if one server in location has full channel, the next viewer from the same geographical point will be balanced to the neighbor server.

And of course, if one server is out of work for any reason, traffic will be re-routed to another server in the same cluster. This means viewers will never get drops in the Live Stream.

No, you don’t.
We have our own CDN. With servers all over the world. You don’t need to look for another provider for your stream’s delivery. It’s all built-in for you.
Our CDN is preset for streaming by default if you’re using our Streaming Platform.
All the settings for caching playlists and chunks are made by us, based on the experience and average streams parameters. We also cache streams not in the HDD/SDD of our servers but in the Operative Memory which means that content will be delivered to viewers faster and with no freezes.

The information will be here soon.

We use our own infrastructure. This means - we can set-up the ingester/transcoder for you in any location where Gcore has Cloud service.
By default, we have ingesters/transcoders in Ashburn (the USA), Luxembourg (to cover Europe) and Singapour (to cover Asia). Of course, they’re united into clusters for redundancy and overcoming huge loads.
Those locations are usually enough for the streamers from the USA/Europe/Asia. And of course, we deliver these streams using our CDN. So, if your streamer is located in the USA and the viewers are in Europe, viewers will get the stream from the European servers.
But If you still want something closer to your streamers, check the map and send a request to us.

We have the following Latency for the following protocols:

  • Reduced Latency HLS (the most common for any cases) – 6-8 sec
  • Low Latency DASH (the most stable LL, but doesn’t work in native iOS AVPlayer, although it works in Safari, iOS, with our player) – 3-4 sec
  • Apple Low Latency HLS (LL working on any device including iOS) – 4 sec
  • HESP (special HTTP-based protocol, works only in the specific player) – less than 1 sec

We also have WebRTC for the case with VideoCalls - real-time latency. But that technology isn’t working with CDN. So, it’s better to use WebRTC for real-time communication, not for high-quality streams. Read more on that here.

We don’t charge for our basic transcoding in the basic protocols (that means quality up to 1080, HLS/DASH/LL-HLS).
So, you pay only for the minutes your viewers watch the stream.
Price for 1 min of watching is Є0.001. For example, you had a 1-hour stream and 10 viewers have been watching it from the beginning till the end. So, 60mins * 10viewers * 0.001Є = 0,6Є
We’re charging per month. That means, if during one month there was only one 1-hour stream with 10 viewers - you pay only for that.
If in the next month you expand to ten 100-mins streams with 100 viewers watching each stream from the beginning till the end, you’ll pay 100users * 100mins * 10streams * Є0.001 = 100Є
So, it depends on your consumption. You don’t pay more than you’ve spent.

Contact us to get personalized offer

Tell us about the challenges of your business, and we’ll help you grow in any country in the world.