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In the English documentation page tutorial.md, there is a link called "supported browsers". It does not link to anywhere. The user just arrives at the list of files in the "en" directory. I think the Japanese documentation has the same issue.

I have the following error when i'm trying to load use Unity Render Streaming :

The version of the hardware codec driver does not support API

I tried to look the dependencies on this website but i didn't found the solution, do you have an idea ? https://developer.nvidia.com/nvidia-video-codec-sdk

Generational GC | Mono
https://www.mono-project.com/docs/advanced/garbage-collector/sgen/

The Mono runtime will automatically register all threads that are created from the managed world with the garbage collector. For developers embedding Mono it is important that they register with the runtime any additional thread they create that manipulates managed objects with mono_thread_attach.

I suspect this statement also applies to Boehm GC. An solution alternative for (and possibly preferred to) mono_thread_attach is to create a thread in a managed code and use rtc::ThreadManager::WrapCurrentThread.

in L.93 of Documentation~/en/https.md , we have "business unity" but it should be "business unit".

This issue reported on the user forum.
https://forum.unity.com/threads/unity-render-streaming-introduction-faq.742481/page-4#post-5430336

Not sure about the issue caused by the unity runtime specification or not.

Hi,
Thanks for this great plugin!

There seems to be a bug in the webapp: on Firefox, at least, e.wheelDelta is undefined in register-event.js:sendMouseWheel(e) :

function sendMouseWheel(e) {
    console.log("mouse wheel with delta " + e.wheelDelta);
// Displays: mouse wheel with delta undefined in dev console

According to this StackOverflow post, e.delta[X/Y/Z] should be used instead of e.wheelDelta.

Note: I'm using the webserver.exe downloaded from Unity.

Thanks!

Unity project is running fine however after downloading the Webapp.exe file, i couldnt open it.

see attached screenshot if helps.

Can I configure this project to use Kurento Media Server?

Hi.
I read the source code of WebApp, and I learned that the signaling server is implemented with http instead of socket.io. May I ask why?
Thanks.

Hi,

I am looking at the FAQ, where it says that "Unity is not supported for rendering server". What does this mean, and where can I get more information about this?

Is there any preventing me from deploying this to a cloud service like Azure or Google Cloud Platform?

--

## Is it available to deploy Unity to cloud service like GCP as a rendering server?

Unity is not supported using for rendering server.

Copied from: https://github.com/Unity-Technologies/UnityRenderStreaming/blob/release/1.1.3/Packages/com.unity.renderstreaming/Documentation~/en/faq.md#is-it-available-to-deploy-unity-to-cloud-service-like-gcp-as-a-rendering-server

Hi, I want to made a videoplayer in unity like the web app sample. after I established ICE connection, I found that RTCPeerConnection.OnTrack event NOT being called.
It seems there might be some issues in Unity.WebRTC.dll ?

could you please check it? thank you.

DllNotFoundException: Packages/com.unity.webrtc/Runtime/Plugins/x86_64/webrtc.dll Unity.WebRTC.WebRTC.Initialize () (at Library/PackageCache/com.unity.webrtc@0f45d9be2546030d6e7c9f71788670ddc2b88554/Runtime/Scripts/WebRTC.cs:224) Unity.RenderStreaming.RenderStreaming.Awake () (at Assets/Scripts/RenderStreaming.cs:59)

This error occurs when checking out the release/1.1.3 branch and launching the project using Unity version 2019.3.0f5.

As a result, the web player does not load the scene. Logs from Chrome (desktop) show:

video-player.js:57 signalingState changed: Event
video-player.js:67 iceGatheringState changed: Event
video-player.js:67 iceGatheringState changed: Event
register-events.js:274 x: NaN, y: NaN, scale: undefined, originX: undefined, originY: undefined mouse button:0
video-player.js:208 Connection not ready

Related package versions are:

• High Definition RP: v7.1.1
• Input System: v0.96
• Unity Render Streaming: v1.1.2
• WebRTC: v1.1.0

Hi there,
I see there is "Add HW Encoder" featured in V2/2.1, but isn't the plugin already using hardware acceleration with NVIDIA cards? Or do you mean add support for AMD too?

Thanks :)

MediaStreamSample passes delegates of instance methods to unmanaged code, which is prohibited on IL2CPP. Maybe you can create System.GCHandles referring to such an delegate and pass them with function pointers of delegates of static methods.

While trying to download webserver.exe from edit->render streaming->Download webapp it downloads webserver.exe with 0 bytes in size

Is there a way to monitor bandwidth consumed by the render streaming web client? The information in the browser network tab is redundant and not very useful as most of them are GET request for candidate and answer

I run webserver.exe as a https server and the render streaming works.
I try to use the url "wss://(my domain name): port" to establish websocket connection in my unity project to this remote server.
However the websocket client can not connect to the remote server.
How can I establish the websocket connection?
I try to create a unity-to-unity render streaming.

In windows OS, very frequently the camera will be positioned in a strange angle upon connected. And it would be very hard to reset the position. It would be good if a button for reset camera being implemented.

Hi, I want connect webserver in unity as the way in broswer, all is OK but the OnTrack handler is not called, I see in broswer it works well. It means, when setting the peer local as offer and remote as answer，OnTrack doesnot work。

If I click on play on the sample scene I get these errors:

DllNotFoundException: Packages/com.unity.webrtc/Runtime/Plugins/x86_64/webrtc.dll
Unity.WebRTC.WebRTC.Initialize (Unity.WebRTC.EncoderType type) (at Library/PackageCache/com.unity.webrtc@b00cafe0bc6fd7170eab00689eaafae2d1698299/Runtime/Scripts/WebRTC.cs:228)
Unity.RenderStreaming.RenderStreaming.Awake () (at Assets/Scripts/RenderStreaming.cs:63)

Assertion failed on expression: 'sceneRanges.empty()'

And streaming is not working.

Enabling the render streaming component stops (or lags) microphone input programmatically added to an audio source on a gameobject. Is this by design? When render streaming is on does it take over control of the microphone? If so, is there a way to use the microphone while render streaming?

2. Does this work outside of the local network? tried on my labtop and it didnt work (because the graphics card was too old), however on my windows server computer it worked. However when I tried to connect to it, it didnt work.
   video-player.js:155 Uncaught (in promise) DOMException: Failed to execute 'addIceCandidate' on 'RTCPeerConnection': Error processing ICE candidate

Latest chrome used
Also latest chrome beta.

What function sends the camera frame to the WebRTC encoder?

It appears that RenderStreaming.cs tries to send every single rendered frame through WebRTC.

If the PC application is running more than 60fps and is viewed in Chrome on an Android device (Samsung Galaxy S8) it tries to decode a 60fps+ video stream on the mobile phone.

The phone cannot handle a stream this fast so it loses a ton of frames/data and datamoshing occurs: http://datamoshing.com/wp-content/uploads/2017/02/Hotline-Bling-1038x576.png

Turning off vsync and setting Application.targetFrameRate to 15 creates a stable video stream.

It would be great to be able to manually call what frames get sent, so that the PC application can run at 60fps but send a 15fps stream if necessary.

When I use Unity Editor or export to .exe, the program runs well.
But when I use c++ CreateProcess API Or vb script

    Set ws = createObject("WScript.shell") 
    ws.run "xxxx.exe",vbhide

to run the program with hidden window, the delay is so high(about 2-3s), that can't interact with program.

Where is the cause of such a high delay?
or
How it run with --batchMode (Silent mode)?

This issue is reproduced by Unity Render Streaming Template version 1.2.1.

It should make other Assembly Difinition for editor script.

I am using the URP example and facing this weird color issue.

I am unable to change camera's target texture setting at runtime since that is not allowed

Ubuntu:18.04
Render streaming version: 1.2.0
Unity: 2019.3.3f1
URP: 7.1.8

I have tried to use the unity asset "Embedded Browser" to receive the render streaming.
This is the asset:
https://assetstore.unity.com/packages/tools/gui/embedded-browser-55459
However I got the error message "Network Error: Unity.WebRTC.RTError." in the scene of this UnityRenderStreaming project.

I run webserver.exe as a https server. It does work that the render streaming shows on different browsers, such as Chrome, Safari at Windows/macOS/iOs/Android devices.

Is it possible to show the render streaming on the embedded browser in the unity scene?

I've ran the test scenes, and all displays properly on my computer. When I test on iOS device (iPad Pro specifically), I do not get any visuals.

Creating an SSL certificate based on your tutorial, gives me errors in all browsers about invalid certificate. I have followed your SSL tutorial 3 times now and each time the same result.

On unity 2019.1.14f1, installed the package from the package manager, the input system dialog box appeared, clicked yes, now am stuck on the import unity package window, with an endless loop of "preparing package: decompressing" and an error "unauthorizedAccessException: Access to the path ...\RenderStreamingHDRPSettings.json is denied"

Hi!
Is it possible to make the streaming work via WiFi without an Internet connection?

i have used chromiumwebbrowser and added "enable-media-stream" params
settings.CefCommandLineArgs.Add("enable-media-stream", "enable-media-stream");
but the video can not displayed.
in RenderStreaming.cs there is a error.
at 187 line
opLocalDesc.isError is true
how can i fixed this error?

I followed this guide to set up HTTPS certificates to run the demo on Safari on iOS:

https://github.com/Unity-Technologies/UnityRenderStreaming/blob/b17d351696643a04bc59cefc4189117e10584b5b/Packages/com.unity.template.renderstreaming/Documentation~/en/https.md

The Webserver starts and I am able to connect to the web app on iOS on port 443.

However in the editor this error appears:

Network Error: Cannot connect to destination host
UnityEngine.Debug:LogError(Object)
Unity.RenderStreaming.<Start>d__15:MoveNext() (at Assets/Scripts/RenderStreaming.cs:87)
UnityEngine.SetupCoroutine:InvokeMoveNext(IEnumerator, IntPtr)

The iOS app never starts streaming.

I have the following error and problem when i'm trying to load use Unity Render Streaming :

First , When I run this scene in unity on my laptop, Windows 10,the graphics card used by my device is NVIDIA GeForce 940MX. The screen of unity is normal, but in the browser, the scene cannot be displayed, which means that the video stream is not coming and the sound is not coming .
Unity has error : The version of the hardware codec driver does not support API ....... It seems that what GitHub said is the hardware problem of the graphics card.
Second, I used another desktop, Windows 7, which had a GTX1070 graphics card. The driver had been updated.This time there was no error, and the browser also does not show the scene.
What's going on?Why can't the browser display the scene?

Original post reported by a user.

(Render Texture version)

(Unity Version)

Versions used

• Unity 2019.2.9f1
• LWRP 6.9.1

Hi,

thanks for this great plugin!

We'd like to stream AR from a high performance edge server to a headset to enable super-high detailed rendering at interactive framerates in AR. Is this something that's on the roadmap, or would it be feasible to adjust the source code of the plugin ourselves to do this?

Environment

• MacOS
• Unity Editor 2019.3.0f6
• Input System 1.0.0-preview.4

How to reproduce the problem

1. Open template project.
2. Open sample scene and play.
3. Switch other application from Unity Editor with the keyboard shortcut (Command+Tab)
4. Switch Unity Editor with the keyboard shortcut (Command+Tab)

Status

This issue caused by "Input System". We are waiting for their answer.

https://forum.unity.com/threads/unity-render-streaming-introduction-faq.742481/page-3#post-5244842

Changes required by a user to avoid crash on non-Windows Platforms:

1. Unity.WebRTC.Runtime.asmdef:
   Exclude non-Windows-Platforms
2. Add some #if UNITY_STANDALONE_WIN || UNITY_EDITOR_WIN to Assets/Scripts/RenderStreaming.cs so that it is considered only for Windows (or to whatever classes you have on your own that uses namespace Unity.WebRTC)

I am trying to test the multicamera2.0 branch but it seems that the VideoStreamTrack file is not added to the project

Play for more than ten hours, the delay is increased to 2 seconds.
Refresh the page, the delay returns to normal.

Some users claimed the issue on Unity Forum what it is not work correctly with 4K streaming.
It might be needed to investigate in detail.

https://forum.unity.com/threads/unity-render-streaming-introduction-faq.742481/page-4#post-5540311
https://forum.unity.com/threads/unity-render-streaming-introduction-faq.742481/page-4#post-5546485
https://forum.unity.com/threads/unity-render-streaming-introduction-faq.742481/page-2#post-5109068

Just wondering if the web server can generate https addresses?

This is Nemo from AssetStore team. First, thanks for your great work. It makes large assets preview possible. We have finished a scalable signal server based on your implementation and there is a possible problem on the c# side.

When calling GetOffer, you are trying to get an array of {connectionId, sdp}. The problem is that sdp might be super long and throw error when more than 2 offers are inside the response list.

In my case, this is ```{"offers":[{"connectionId":"b178a404-6915-4ca9-bb5c-101725df98d4","sdp":"v=0\r\no=- 8883974332256265528 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1 2\r\na=msid-semantic: WMS\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:1cjg\r\na=ice-pwd:Vsc9XkzFm6mZfjDI5T9blehc\r\na=ice-options:trickle\r\na=fingerprint:sha-256 CB:FB:C8:86:3E:05:D7:A2:BD:11:C4:2B:5D:9B:1A:DB:1B:5B:E5:E3:BC:3F:56:C6:03:C1:B3:B4:43:71:D0:52\r\na=setup:actpass\r\na=mid:0\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1;stereo=1;maxaveragebitrate=1048576\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 122 127 121 125 107 108 109 124 120 123 119 114 115 116\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:1cjg\r\na=ice-pwd:Vsc9XkzFm6mZfjDI5T9blehc\r\na=ice-options:trickle\r\na=fingerprint:sha-256 CB:FB:C8:86:3E:05:D7:A2:BD:11:C4:2B:5D:9B:1A:DB:1B:5B:E5:E3:BC:3F:56:C6:03:C1:B3:B4:43:71:D0:52\r\na=setup:actpass\r\na=mid:1\r\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:13 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:12 urn:3gpp:video-orientation\r\na=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07\r\na=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\na=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=fmtp:98 profile-id=0\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:100 VP9/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 profile-id=2\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:122 rtx/90000\r\na=fmtp:122 apt=102\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\r\na=rtpmap:121 rtx/90000\r\na=fmtp:121 apt=127\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 H264/90000\r\na=rtcp-fb:108 goog-remb\r\na=rtcp-fb:108 transport-cc\r\na=rtcp-fb:108 ccm fir\r\na=rtcp-fb:108 nack\r\na=rtcp-fb:108 nack pli\r\na=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f\r\na=rtpmap:109 rtx/90000\r\na=fmtp:109 apt=108\r\na=rtpmap:124 H264/90000\r\na=rtcp-fb:124 goog-remb\r\na=rtcp-fb:124 transport-cc\r\na=rtcp-fb:124 ccm fir\r\na=rtcp-fb:124 nack\r\na=rtcp-fb:124 nack pli\r\na=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032\r\na=rtpmap:120 rtx/90000\r\na=fmtp:120 apt=124\r\na=rtpmap:123 H264/90000\r\na=rtcp-fb:123 goog-remb\r\na=rtcp-fb:123 transport-cc\r\na=rtcp-fb:123 ccm fir\r\na=rtcp-fb:123 nack\r\na=rtcp-fb:123 nack pli\r\na=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032\r\na=rtpmap:119 rtx/90000\r\na=fmtp:119 apt=123\r\na=rtpmap:114 red/90000\r\na=rtpmap:115 rtx/90000\r\na=fmtp:115 apt=114\r\na=rtpmap:116 ulpfec/90000\r\nm=application 9 UDP/DTLS/SCTP webrtc-datachannel\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:1cjg\r\na=ice-pwd:Vsc9XkzFm6mZfjDI5T9blehc\r\na=ice-options:trickle\r\na=fingerprint:sha-256 CB:FB:C8:86:3E:05:D7:A2:BD:11:C4:2B:5D:9B:1A:DB:1B:5B:E5:E3:BC:3F:56:C6:03:C1:B3:B4:43:71:D0:52\r\na=setup:actpass\r\na=mid:2\r\na=sctp-port:5000\r\na=max-message-size:262144\r\n"},{"connectionId":"93e02d39-30ad-4055-8cab-4ec526a9fe2e","sdp":"v=0\r\no=- 2047041661280945092 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1 2\r\na=msid-semantic: WMS\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:ksu1\r\na=ice-pwd:wjc6Dsg3h5MP833Fr4s1HePH\r\na=ice-options:trickle\r\na=fingerprint:sha-256 0A:C1:78:92:7D:5A:8A:AC:60:F8:FB:C1:3D:07:21:5D:21:B3:5C:F5:19:7E:36:A0:94:0C:1C:F8:FF:06:27:D2\r\na=setup:actpass\r\na=mid:0\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1;stereo=1;maxaveragebitrate=1048576\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 122 127 121 125 107 108 109 124 120 123 119 114 115 116\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:ksu1\r\na=ice-pwd:wjc6Dsg3h5MP833Fr4s1HePH\r\na=ice-options:trickle\r\na=fingerprint:sha-256 0A:C1:78:92:7D:5A:8A:AC:60:F8:FB:C1:3D:07:21:5D:21:B3:5C:F5:19:7E:36:A0:94:0C:1C:F8:FF:06:27:D2\r\na=setup:actpass\r\na=mid:1\r\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:13 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:12 urn:3gpp:video-orientation\r\na=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07\r\na=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\na=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transpor
UnityEngine.Debug:LogError(Object)
Unity.RenderStreaming.DownloadHandlerJson`1:ReceiveData(Byte[], Int32) (at Assets/Scripts/Signaling.cs:34)

I think there are 2 possible way to fix.

 - Get array of {connectionId} for the first time. And check if alrealy handled, then get a single offer containing sdp back in one call. In this way, we would make two http calls.
 - Change GetOffer of returning array to return a single offer object. Since you are looping to get offer. It might be the best way.

Our team has rewritten a scalable version of signal server in Go. This issue would be fixed using the second way. But related logic will be diverged from yours. FYI

Since HDRP 7.1.2 has a critical bug that caused fail test with upm-ci, template testing on CI is disabled. This bug was already fixed and release soon as HDRP 7.2.

This log is for the problem.

Culling group was not disposed. You have to call Dispose explicitly from the main thread. This will likely result in a crash.

Related issue on the Issue Tracker
https://issuetracker.unity3d.com/issues/hdrp-culling-group-was-not-disposed-message-is-shown-in-editor-dot-log-after-exiting-a-project

Hi,

I tried to use Unity UI with RenderStreaming, with all types of UI (screen overlay, world, world overlay) and it does not respond to click event received from WebRTC.

Do you know if there is a special setting to make it work, or if it's simply not possible with the current implementation?

I looked into the new Input system documentation but had no luck so far.

Thanks for your help. :)

hi Sindharta,

As discussed over email, latest Chrome doesnt load content.

please see attached screenshot of error message and a screenshot when content plays well in unity window but nothing on Chrome web page.

Hello,
I have an issue when trying to connect as a receiver on Chrome/Android. The connection doesn't seem to get established. That's what the Unity console tells me on my Windows machine:

Network Error: Unity.WebRTC.RTCError
UnityEngine.Debug:LogError(Object)
Unity.RenderStreaming.d__20:MoveNext() (at Assets/Scripts/RenderStreaming.cs:183)
UnityEngine.SetupCoroutine:InvokeMoveNext(IEnumerator, IntPtr)

Nothing specific appears in the webserver.exe console apart from the IP addresses of the broadcasting device.

The connection works fine with Chrome/MacOS as a receiver. I'm using the preview release version 1.2.3
Do you have any idea where the issue might come from?
Thanks in advance.