SRT in GStreamer
SRT, the open source video transport protocol that enables the delivery of high-quality and secure, low latency video, has been integrated into GStreamer.
By Olivier Crête, Multimedia Lead at Collabora.
Transmitting low delay, high quality video over the Internet is hard. The trade-off is normally between video quality and transmission delay (or latency). Internet video has up to now been segregated into two segments: video streaming and video calls. On the first side, streaming video has taken over the world of the video distribution using segmented streaming technologies such as HLS and DASH, allowing services like Netflix to flourish. On the second side, you have VoIP systems, which are generally targeted a relatively low bitrate using low latency technologies such as RTP and WebRTC, and they don't result in a broadcast grade result. SRT bridges that gap by allowing the transfer of broadcast grade video at low latencies.
The SRT protocol achieves these goal using two techniques. First, if a packet is lost, it will retransmit it, but it will only do that for a certain amount of time which is determined by the configured latency, this means that the latency is bounded by the application. Second, it tries to guess the available bandwidth based on the algorithms from UDT, this means that it can then avoid sending at a rate that exceeds the link's capacity, but it also makes this information available to the application (to the encoder) so that it can adjust the encoding bitrate to not exceed the available bandwidth ensuring the best possible quality. Using the combination of these techniques, we can achieve broadcast grade video over the Internet if the bandwidth is sufficient.
At Collabora, we're very excited with the possibilities created by SRT, so we decided to integrate it into GStreamer, the most versatile multimedia framework out there!