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Changed webRTC jitterbuffer to 1.5s for clients receiving streams, updated WebRTC text to reflect recent TURN(S) capabilities.

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rainbow napkin 2026-10-09 20:11:05 -04:00
parent 0968a5eba6
commit 881dcc2e9c
1 changed files with 7 additions and 7 deletions
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14
www/js/channel/webRTCHandler.js
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@ -164,6 +164,12 @@ class webRTCHandler{
//Set offer as peerConnections local description
await this.currentStream.setLocalDescription(offer);
//For each receiver from the current streamer
for(let receiver of this.currentStream.getReceivers()){
//Set jitter buffer target to 1.5s, since we care more about consistent video quality than we do latency
receiver.jitterBufferTarget = 1500;
}
//Send RTC connection offer to server to be relayed to streamer's device
this.client.socket.emit("relayRTCData", {type:"request",body:offer});
}
@ -289,17 +295,11 @@ class webRTCHandler{
title.innerText = "WebRTC Error";
let text = document.createElement('p');
text.innerText = "Looks like we're having issues connecting you to your stream. Make sure you're not behind a CGNAT or something.";
let link = document.createElement('a');
link.innerText = "Click here for more info...";
link.href = "/streamInfo";
link.target = "_blank";
text.innerHTML = "Looks like we're having issues connecting you to your stream.<br><br>If you can't get it after a refresh, try again later or on another connection/device.";
let textBlock = document.createElement('div');
textBlock.appendChild(title);
textBlock.appendChild(text);
textBlock.appendChild(link);
let errorBlock = document.createElement('div');
errorBlock.id = "web-rtc-error-block"