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Live stream best practices

Starting a live stream

  • When a stream is starting it might take up to 30 seconds for the playback to start delivering.
  • Before starting the stream, it is important to have an intro (static picture, countdown and etc) to the beginning of the stream that will allow users to connect before the stream starts.
  • In order to prevent the live stream from getting stuck (buffering indefinitely), please ensure that you are following the recommended settings.
  • Make sure to verify that the connection speed is adequate and stable before the stream is starting.

Streaming protocols supports both RTMP and SRT protocols for live streaming. Accepting live streams through SRT is currently in beta - reach out if you have any questions or encounter any issues!

SRT details

  • The minimum accepted value for latency is 120 ms, lower values will be ignored.
  • The required SRT protocol version should be 1.3 or higher - check your broadcasting application to make sure it supports the correct version.

Which protocol to use

Take network connection into consideration - if you or your users stream in stable conditions with a low risk of congestion, RTMP will probably achieve a better latency than SRT. If you or your users experience lots of buffering, or the network you use is known to be unstable, or network conditions may change like when using a mobile network, then SRT may be a good fit.

Rembember that both protocols have their pros and cons - with bad network conditions, viewers might encounter buffering with RTMP, or they might encounter audio or visual glitches when using SRT.

DVR - reading history in live streams

When a live stream is ongoing, viewers can replay earlier content with the DVR feature. The available buffer is 1 hour, and is only available in live streams that are actively broadcasting.

When the stream ends, stores the stream for another 4 to 5 minutes, subsequently ending the stream and disposing of the cached video.

Ending the live stream

Make sure to use an outro (static picture, music and etc.) at the end of the stream for 30 seconds. This is a good way for users to indicate that the stream has ended.

When reaching the end of the playback, the player buffers until the live stream is deleted, which could take from 10 seconds to 5 minutes and could cause the live stream not to end properly even if you have stopped the broadcast.

Connectivity loss reconnection

Reconnection is handled by, however, an edge case might occur (very slim chance), which will result in the inability to reconnect to the stream with good quality. In this case, we recommend creating a manual stream reconnection, where the streamer will create a new streaming key while the consumers will have to refresh their player instance.

Streaming servers

ProtocolDescriptionServer URL
RTMPThe default streaming server.rtmp://
SRTThe SRT server, currently in{stream_key}

Recommended setting for ingestion

Video codec: H.264
Audio codec: AAC/MP3
Bitrate encoding: CBR
Keyframe Interval: 2 second

QualityFramerateVideo bitrateAudio sample rateAudio Bitrate
240p25-30 fps300-700 Kbps44,1 kHz64 Kbps
360p25-30 fps400-1000 Kbps44,1 kHz128 Kbps
480p25-30 fps500-2000 Kbps44,1 kHz128 Kbps
720p(30 fps)25-30 fps 1500-4000 Kbps44,1 kHz128 Kbps
720p(60 fps)60 fps 2250-6000 Kbps44,1 kHz128 Kbps
1080p(30 fps)25-30 fps 3000-6000 Kbps44,1 kHz128 Kbps
1080p(60 fps)60 fps 4500-9000 Kbps44,1 kHz128 Kbps
2160(4K)(30 fps)25-30 fps 13000-34000 Kbps44,1 kHz192 Kbps
2160(4K)(60 fps)60 fps 20000-51000 Kbps44,1 kHz192 Kbps


  • When using the sandbox environment, live streaming is limited to 24 hours.
  • The video codec must be H.264
  • The audio codec must be AAC or MP3
  • DVR is exactly 1 hour
  • During a disconnection on ingest side, re-connection must occur within 10 seconds
  • When creating a new live event using the same streamID without keeping the DVR of a previous live, you must wait at least 5 minutes before re-using it

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