Packet loss is a situation where data traveling across a network fails to reach its intended destination. Packet loss is often measured as a ratio of the number of lost packets compared to the total number of packets sent. When data needs to be delivered reliably, packets are usually retransmitted, but this can increase network latency.
There are a number of reasons why packet loss may occur from bandwidth or device limitations to firewalls that block data transfers, but the most common cause of packet loss is network congestion. That’s because packet loss is often used by the transmission control protocol (TCP) to reduce network congestion and increase the throughput for connections.
Packet loss during video streaming is a problem because it negatively impacts the viewing experience. For example, if all of the packets containing the data for a particular section of video is lost, issues like pixelated images, video content gaps, and even total playback failure can occur. That’s why broadcasters use several methods to reduce packet loss during video streaming.
When it comes to livestreaming, many broadcasters often choose streaming protocols that use Forward Error Correction (FEC) techniques to protect against packet loss. This involves sending redundant data to plan for inevitable packet loss, but doesn’t always offer sufficient redundancy for large chunks of packet loss. FEC is most effective for packet loss that’s less than 3-5%.
Automatic Repeat Request (ARQ) is another technique that’s more efficient than FEC because it only retransmits the packets that haven’t been confirmed as received. This helps ensure reliable data transmission over unreliable networks, but can still cause traffic congestion when there’s a high rate of packet loss.
Many online video platforms also use RTMP for livestream ingestion, which cannot overcome excessive packet loss. That means incomplete data reaches the online video platform before it’s even processed for distribution and playback. Additional packet loss during delivery to end-users can further degrade the quality of the livestream as well.
Latency the time it takes for data to reach its destination—is closely related to packet loss as well. While there are numerous ways that latency is introduced during video streaming workflows, such as a slow encoding or transcoding process, transmission and delivery can also cause latency.
For example, when dropped packets are retransmitted, there is a delay until the streaming protocol can receive the complete video content. Retransmitting packets also use additional bandwidth, which can cause further packet loss or latency as the video stream continues. That means broadcasters need to balance packet loss and latency for more effective video streaming.
Resi’s Resilient Streaming Protocol (RSP) is designed to protect against quality loss using selective retransmission and data verification checks. If RSP detects that incomplete video data was received, the protocol automatically retransmits the corrupted data to ensure a perfect copy of the video file reaches its destination. That means RSP can also overcome packet loss, bandwidth limitations, network outages, and other situations that negatively impact video streaming.
