Network Performance

Bitrate

Bitrate defines the amount of video and audio data sent by the local peer to remote peers per second, measured in bits per second (bps). The maximum achievable outgoing bitrate depends on factors like:

• RTT, packet loss, and available network bandwidth.
• Bitrate values configured for the respective role under templates.

Bitrate values

• Standard bitrate value for audio is 32kbps; for video it can be configured under the template settings.
• Higher bitrate value means more data per second is being sent, allowing for better video/audio quality. But if the bitrate exceeds network capacity it leads to packet loss and quality degradation.

Frame Rate (FPS)

The frame rate refers to the number of video frames transmitted per second by the local peer to the remote peers in a video call.

Frame rates and video quality

• Higher outgoing frame rates result in smoother, less choppy video quality at the receiver's end. However, high frame rates require higher bitrates.
• Lower outgoing frame rates allow the local peer to reduce their transmitted bitrate while maintaining visual quality.

Packet Loss

Packet loss occurs when media packets being sent by the local peer fail to reach the remote peers, resulting in degraded audio/video quality of the peer. It occur due to network congestion, insufficient bandwidth, faulty network hardware, etc.

Measuring Packet loss

• Some amount of data loss is inevitable, so the severity is measured by the percentage of total packets lost.
• Packet loss below 5% is generally considered acceptable and normal.
• Beyond 5%, high outgoing packet loss can significantly degrade the video resolution, frame rate, and image quality observed by remote viewers. Audio can be jittery or broken up or may miss sections of audio.

Round Trip Time (RTT)

Measures the time in milliseconds (ms) it takes for a packet of data to travel from the source(client) to the destination and back again. Higher the RTT, higher the latency.

Jitter

Jitter refers to the variation in delay between delivered media packets being sent from the local peer to the remote peers. High outgoing jitter can cause received video to stutter or display jerky movements and make audio sound choppy or distorted to remote viewers.

Quality Limitations

There are two main limitations that can severely impact the outgoing media quality transmitted from a local peer to remote peers:

CPU Limitations

Insufficient CPU cores or processing power can cause delays in encoding and transmitting outgoing audio/video packets. This leads to increased latency, jitter, and potential packet loss for the streams received by remote peers.

Network Limitations

Bandwidth, throughput, or network equipment constraints can create congestion and bottlenecks specifically along the local peer's upload path. This results in higher outgoing latency, jitter, and packet loss that significantly reduces the media quality observed by remote participants.

Bitrate

When incoming media quality, bitrate is defined as the sum of all the bitrates published by all the remote peers in the call. It is defined using:

• Expected Bitrate - The bitrate a client expects to receive based on the video resolution and layers selected according to the call template settings and the number of tracks being downloaded. This is configured for the role under templates.
  
• Actual Bitrate - The bitrate actually received by the client, limited by the sender's upload bandwidth and network conditionsanddownlink network conditions of the peer`
  

CQS and Bitrate

• Connection quality score uses the ratio of Actual Bitrate received/Expected Bitrate to determine its video and audio quality.
• In ideal scenarios, the actual bitrate received should be equal to expected bitrate. If there's a delta in actual and expected bitrate then there's a drop in video quality consumed by the peer

Round Trip Time

Measures the time in milliseconds (ms) it takes for a video or audio packet to travel from the remote peer to the local peer and back to the remote peer.

High outgoing RTT reduces interactivity and introduces lag to remote users.

Packet Loss

Packet loss occurs when media packets transmitted from remote peers fail to reach the local peer. This results in degradation of the incoming audio/video quality received by the local peer. Some potential root causes include Network congestion, Insufficient bandwidth, Faulty network hardware

Packet loss on incoming media quality:

• Minor packet loss is inevitable, so the severity is measured as a percentage of total packets expected to arrive vs packets lost.
• Up to 5% loss is generally acceptable without noticeable quality impact.
• Beyond 5%, high incoming packet loss can significantly lower the video resolution, frame rate, and image quality received by the local peer. Incoming audio can sound broken, intermittent, or have missing sections.