Despite advances in internet streaming, live video transport still faces latency challenges due to:

  • Packet loss when sending UDP-based low latency MPEG Transport Streams over the public internet
  • Bandwidth limitations introduced by network congestion control when using the TCP protocol in RTMP
  • Segment-based protocols such as HLS and DASH transmitted over HTTP (TCP)

Generally, overcoming these problems requires the provisioning of high-cost reserved links like MPLS or satellite networks. With the open source implementation of SRT, any developer has access to a streaming protocol that provides a secure and reliable solution for low latency video transport with packet loss recovery, end-to-end security with AES encryption, network health monitoring between endpoints, and simplified firewall traversal. Furthermore, developers can enable SRT functionality over any network, including cost-effective public internet.

SRT is a video streaming technology that leverages a combination of broad video ecosystem components and technology advantages to enable the best quality live video over unpredictable networks, even the public internet. SRT accounts for packet loss, jitter, and fluctuating bandwidth, maintaining the integrity and quality of your video. SRT enables you to keep your streams secure and easily traverse firewalls.

SRT source code is freely available on GitHub at https:/ Developers can improve upon, use, and re-contribute to SRT under the MPL-2.0 license.