For network streaming applications, it is always important to keep in mind that network capabilities may vary depending on multiple factors. These variations might affect the streaming quality and performance if it is not properly managed to change dynamically based on network changes.
RidgeRun created an element called GstRtpNetCc with capabilities to analyze the RTP stream and provide feedback to user applications with the maximum stream bitrate the application should set to avoid packet loss and streaming problems.
This element uses a Network Congestion Control Algorithm from Google to provide the bitrate feedback. WebRTC applications can use this feedback to provide REMB or Transport-CC RTCP messages to the peer to control the streaming bitrate dynamically. RidgeRun GstWebRTC has implemented this plugin into its design.
Learn more at our developer wiki.Platforms Supported
- X86
- Arm - IMX6
- Arm - NVIDIA Jetson
Coming soon..