Introduction to SRT

Secure Reliable Transport (SRT) is an open source transport technology that optimizes streaming performance across unpredictable networks, such as the Internet.

SRT is applied to contribution and distribution endpoints as part of a video stream workflow to deliver the best quality and lowest latency video at all times.

As audio/video packets are streamed from a source to a destination device, SRT detects and adapts to the real-time network conditions between the two endpoints. SRT helps compensate for jitter and bandwidth fluctuations due to congestion over noisy networks, such as the Internet. Its error recovery mechanism minimizes the packet loss typical of Internet connections. And SRT supports AES encryption for end-to-end security, keeping your streams safe from prying eyes.

SRT has its roots in the UDP-based Data Transfer (UDT) protocol. While UDT was designed for high throughput file transmission over public networks, it does not do well with live video. SRT is a significantly modified version that supports live video streaming.

Low latency video transmission across IP based networks typically takes the form of MPEG-TS unicast or multicast streams using the UDP protocol. This solution is perfect for protected networks, where any packet loss can be mitigated by enabling forward error correction (FEC). Achieving the same low latency between sites in different cities, countries or even continents is more challenging. While it is possible with satellite links or dedicated MPLS networks, these are expensive solutions. The use of cheaper public internet connectivity, while less expensive, imposes significant bandwidth overhead to achieve the necessary level of packet loss recovery.

Even though UDT was not designed for live streaming, its packet loss recovery mechanism provided an interesting starting point. The original version of SRT included new packet retransmission functionality that reacted immediately to packet loss to enable live streaming.

To achieve low latency streaming, SRT had to address timing issues. The characteristics of a stream from a source network are completely changed by transmission over the public internet, which introduces delays, jitter, and packet loss. This, in turn, leads to problems with decoding, as the audio and video decoders do not receive packets at the expected times. The use of large buffers helps, but latency is increased.

SRT includes a mechanism that recreates the signal characteristics on the receiver side, dramatically reducing the need for buffering. This functionality is part of the SRT protocol itself, so once data comes out of an SRT connection on the receiver side, the stream characteristics have been properly recovered.

Initially developed by Haivision Systems Inc., the SRT protocol was released as open source in April 2017 in partnership with Wowza Media Systems Inc. Open source SRT is distributed under MPL-2.0, which was chosen because it strikes a balance between driving adoption for open source SRT, while encouraging contributions to improve upon it by the community of adopters. Any third party is free to use the SRT source in a larger work regardless of how that larger work is compiled. Should they make source code changes, they would be obligated to make those changes available to the community.

In May 2017, Haivision and Wowza founded the SRT Alliance (www.srtalliance.org), a consortium dedicated to the continued development and adoption of the protocol.

In March 2020, the SRT Protocol RFC was submitted to the IETF as a draft.