Optimize for Low Bandwidth Networks - React Native

When participants join a call from a weak or unstable network, the available bandwidth may not be enough to support the call at its default media settings. VideoSDK provides several controls that you can use to adapt the experience to changing network conditions.

This guide covers recommended approaches for managing the media each participant sends and receives, including choosing the video quality to publish, controlling subscriptions, and keeping participants informed as conditions change. These options can help you reduce bandwidth usage and maintain a reliable call experience.

Before the Call

Measure the participant's connection before media is published, then use the result to choose a starting configuration that suits the connection.

Test the Connection

• Run runPreCallTest() before the participant joins the meeting. It returns a quality score from 1 (BAD) to 5 (EXCELLENT) for both uplink and downlink, along with a factors array that identifies the conditions affecting the score. Use these results to choose a starting media configuration that is appropriate for the participant's connection.

• See Precall Setup for the parameters accepted by runPreCallTest(), the complete result object, and the recommended configuration for different scores and factors.

tip

Evaluate uplink and downlink separately, and use the score as an overall measure of connection quality rather than as an indication of available bandwidth. A low score can result from factors such as high latency even when sufficient bandwidth is available. In that case, lowering video quality may not improve the connection.

Configure the Meeting

Use the following meeting settings to control how participants publish and receive media. Select the settings based on the participant's network conditions and the requirements of your application.

Configure Published Media

• Use customCameraVideoTrack and customMicrophoneAudioTrack to pass media tracks configured for the participant's connection, based on your use case and the pre-call test results. If these are not provided, VideoSDK uses its default track configuration.
• Create the camera track with createCameraVideoTrack() and the microphone track with createMicrophoneAudioTrack(). Both accept an encoderConfig, while the camera track also accepts bitrateMode and maxLayer.
• For constrained connections, consider bitrateMode: "bandwidth_optimized" for the camera to reduce bandwidth usage at the selected resolution. See Optimize Video Tracks for the bitrate used by each video configuration and Optimize Audio Track for the available audio configurations.
• You can also configure the video codec to suit your application's requirements. Codec support and configuration options vary, so see Video Codecs before changing the codec or using it with other video settings.
• To enforce an upper limit on the amount a participant can send, set Maximum Send Bitrate Per Participant for the API key in the VideoSDK Dashboard. The limit applies to every session using that API key and overrides higher bitrate settings configured in your application. See Optimize Audio and Video Usage for details.

Keep Simulcast Enabled

• Keep multiStream enabled, which is the default. With multiStream: true, the camera publishes multiple video quality layers. VideoSDK can then deliver an appropriate layer to each participant based on the receiving conditions and requested video quality.
• With multiStream: false, the camera publishes a single video layer at the configured resolution. All participants then receive that same published layer, even when their network conditions or video layouts differ.
• See Simulcast (Adaptive Bitrate) for more information.

Control Media Consumption

• For large calls, consider autoConsume: false when the application needs explicit control over which streams each participant receives. With this setting, streams are consumed only when requested, allowing the application to subscribe only to the participants required by the interface.
• Use consumeWebcamStreams(), consumeMicStreams(), stopConsumingWebcamStreams(), and stopConsumingMicStreams() on useParticipant as participants enter and leave the visible part of the interface.

Choose the Participant's Media Mode

Use the mode setting to control whether a participant publishes and receives media.

Mode	Description
SEND_AND_RECV	Publishes and receives media. This is the default.
RECV_ONLY	Receives media without publishing. Consider this for participants who only need to watch or listen, or when upload bandwidth is limited.
SIGNALLING_ONLY	Does not publish or receive media. Consider this when the participant does not need real-time media.

• Choose a mode that matches the participant's role and network conditions.
• Use changeMode() to switch between modes during the meeting.

Improve the In-Call Experience

Network and media conditions can change during a meeting. Use VideoSDK events and statistics to keep participants informed when the connection is reconnecting, video quality changes, or media conditions affect their experience.

Keep Participants Informed

Use call and media events to provide clear feedback when conditions change during the meeting. Depending on your application, this can be a toast, banner, status indicator, or other in-call message.

Signal	What it tells you	Recommended response
onQualityLimitation	Reports a limitation affecting the local participant's outgoing media. The type can be bandwidth, congestion, or cpu, and the state can be detected or resolved. See Monitoring Network Quality.	When a bandwidth or congestion limitation is detected, show it to the participant so they can switch to a better network.
onMeetingStateChanged	Indicates a change in the meeting connection state, including RECONNECTING when the connection is interrupted. See Meeting Connection Events.	Show a clear reconnecting message or status indicator so the participant knows the connection is being restored.
onStreamStateChanged	Reports the state of a remote participant's video or screen-share stream as active, freeze-detected, freeze-resolved, stuck, or ended. See Stream Events.	Show feedback on the participant's tile while their stream is frozen or stuck, so the interruption is explained rather than appearing as a broken tile. Clear the feedback when the stream reports freeze-resolved or active.
onVideoQualityChanged	Indicates that a participant has switched to a different simulcast layer. See Simulcast (Adaptive Bitrate).	When relevant to your interface, show a video-quality indicator or other feedback so the participant understands that the received video quality has changed.
getVideoStats() and getAudioStats()	Provide statistics for a remote participant's incoming media, including values such as bitrate, rtt, jitter, and packetsLost. See Understanding Call Quality.	Use these statistics when your application needs to identify or display more detailed information about the quality of a remote stream.
getTransportStats()	Reports the total bitrate being sent and received by the participant.	Use this when your application needs additional information about overall media usage or to support custom quality-related UI.

Optimize Incoming Video

Incoming video is usually the largest part of a participant's bandwidth usage, and a participant only benefits from the quality their layout can actually display. Video is rendered with RTCView, which does not adjust the requested quality on its own, so match the incoming video to each tile yourself using the controls below.

Recommendation	How
Control quality yourself	Use setQuality() to request the quality appropriate for each tile. See Selecting the Video Quality Layer.
Pause streams that are not visible	Use pause() while a participant is outside the visible area and resume() when they return. See Grid with Pagination.
Reduce screen-share quality	Use setScreenShareQuality("low") when bandwidth is limited. Screen-share quality is managed separately from camera video.
Enable adaptive subscriptions	Use enableAdaptiveSubscription() to let VideoSDK prioritise streams for you. Under bandwidth constraints it pauses muted participants' video first, then the least dominant speakers, and keeps pinned participants at the highest quality. See Scalability for Large Participant.

Optimize Data Transfers

Call media is not the only traffic using the participant's connection. Image captures and application updates also consume bandwidth, so keeping these transfers efficient can help preserve bandwidth for the call, especially on weak networks.

Capture Images Efficiently

• captureImage() captures a frame from the local participant's camera at its capture resolution. It can only be called on the local participant and requires the camera to be enabled, so the image always comes from the camera itself rather than from a received video layer.
• For use cases such as document capture or identity verification, this means the image quality does not depend on the published video quality. There is no need to raise the published quality in order to obtain a higher-resolution image.

Consider the following when transferring a captured image:

Recommendation	How
Request the capture from the camera owner	Publish a request on a pubSub topic, then capture the image locally on the device that receives it. See Image Capturer.
Capture only the required size	Pass width and height to captureImage() when a smaller image is sufficient. If the requested size exceeds the camera's capture resolution, the camera resolution is used instead.
Transfer large images carefully	Send the image over pubSub as a Base64 string. For larger images, split the data into chunks and include a transfer ID, chunk index, and total number of chunks so the receiver can reassemble it. See Overview.

Optimize Message Delivery

Chat, reactions, application state, and other updates can also use bandwidth during a call. Choose the delivery method based on how important the data is and how quickly it becomes outdated.

Use case	Recommendation	Guide
Persisted messages delivered when a participant joins	Set oldMessageLimit on usePubSub() when the full message history is not required, and newMessageLimit to cap how many realtime messages are delivered every 500 ms.	Overview
High-frequency realtime topics where stale updates have little value	Set realtimeOverflow: "drop" so new updates are not blocked by messages waiting to be delivered.	Overview
Updates that should survive a brief delay	Keep realtimeOverflow: "queue", the default, and set maxQueue so a subscriber that falls behind can catch up. Anything lost beyond the queue is reported through onMessageDrop.	Overview
Messages that do not need to reach late joiners	Leave persist disabled when publishing, which is the default, so the message is not stored and replayed to participants who join later.	Overview

API Reference

The API references for all the methods and events utilized in this guide are provided below.

• runPreCallTest()
• createCameraVideoTrack()
• createMicrophoneAudioTrack()
• customCameraVideoTrack
• customMicrophoneAudioTrack
• autoConsume
• multiStream
• mode
• changeMode()
• enableAdaptiveSubscription()
• onQualityLimitation
• onMeetingStateChanged
• setQuality()
• setScreenShareQuality()
• captureImage()
• getVideoStats()
• getAudioStats()
• getTransportStats()
• consumeWebcamStreams()
• stopConsumingWebcamStreams()
• consumeMicStreams()
• stopConsumingMicStreams()
• onStreamStateChanged
• onVideoQualityChanged
• pause()
• resume()
• usePubSub