# Does mediasoup support H264 streaming to chrome browser?

**URL:** <https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072>\
**Category:** mediasoup libraries\
**Created:** [July 23, 2021, 4:48am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072 "2021-07-23T04:48:39Z")\
**Posts on this page:** 20\
**Page:** 1

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 23, 2021, 4:48am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/1 "2021-07-23T04:48:39Z")

</div>

I’ve confirmed that chrome can play H264 video via WebRTC (using Janus server), no special config needed(audio still using Opus encoding).

Does mediasoup support H264 streaming? There is no detailed documentation about that.

Please help clarify this. thanks!

---

<div class="post-metadata">

**Author:** ![ibc](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/ibc/32/1540_2.png) [@ibc](https://mediasoup.discourse.group/u/ibc)\
**Post date:** [July 23, 2021, 6:18am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/2 "2021-07-23T06:18:26Z")

</div>

> [@chenzx](#):
>
> Does mediasoup support H264 streaming? There is no detailed documentation about that.

Yes there is

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 23, 2021, 9:13am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/3 "2021-07-23T09:13:48Z")

</div>

Hi, ibc  
i tried with the below config:

```auto
{
    mimeType : 'video/H264',
    clockRate : 90000,
    payloadType : 96,
    parameters : {
        "packetization-mode" : 1,
        "profile-level-id": "640032",
        "x-google-start-bitrate" : 1000,
        "level-asymmetry-allowed" : 1
    },
    rtcpFeedback : [
        {
        "type" : "nack",
        "parameter" : "pli"
        },
        {
        "type" : "ccm",
        "parameter" : "fir"
        },
        {
        "type" : "goog-remb",
        "parameter" : ""
        },
        {
        "type" : "transport-cc",
        "parameter" : ""
        }
    ], // FFmpeg does not support NACK nor PLI/FIR. ?
}

```

This can support chrome’s page2page H264 streaming with 640\*480 size.

But when i use the ffmpeg RTP push command(video only):

```auto
   const callRTPStreamPushCmd = `ffmpeg \
    -re \
    -v info \
    -stream_loop -1 \
    -i ${input_mp4_filepath} \
    -an \
    -c:v copy \
    -f rtp \
    -sdp_file video.sdp \
    "[select=v:f=rtp:ssrc=22222222:payload_type=96]rtp://127.0.0.1:${ports.videoRtp}?rtcpport=${ports.videoRtcp}"`

```

Failed.

And also tried using gstreamer: (modified according to [mediasoup-demo/gstreamer.sh at v3 · versatica/mediasoup-demo · GitHub](https://github.com/versatica/mediasoup-demo/blob/v3/broadcasters/gstreamer.sh))

```auto
   const callRTPStreamPushCmd = `gst-launch-1.0 \
	rtpbin name=rtpbin \
	filesrc location=${input_mp4_filepath} \
	! qtdemux name=demux \
	demux.video_0 \
	! queue \
	! decodebin \
	! rtph264pay pt=96 ssrc=22222222 picture-id-mode=2 \
	! rtpbin.send_rtp_sink_0 \
	rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=${ports.videoRtp} \
	rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=${ports.videoRtcp} sync=false async=false`

```

Also failed. How can i do a H264 RTP stream push with ffmpeg or gstreamer?

---

<div class="post-metadata">

**Author:** ![nazar-pc](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/nazar-pc/32/369_2.png) [@nazar-pc](https://mediasoup.discourse.group/u/nazar-pc)\
**Post date:** [July 23, 2021, 9:31am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/4 "2021-07-23T09:31:12Z")

</div>

You need to configure `profile-level-id` to be the same as the video you’re using AND browser in question should support that particular profile. It is only really safe to use constrained baseline profile if you want H264 to work everywhere.

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 23, 2021, 9:41am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/5 "2021-07-23T09:41:56Z")

</div>

Hi nazar-pc,

How can i know which is the correct `profile-level-id` with my test input mp4 file? (i don’t want to use any transcoding at client side) and how can i know if chrome browser’s WebRTC+H264 stack supports this profile?

My test input mp4 file is 1080p with aac audio encoding(audio is not used here), is inout resolution the reason?

---

<div class="post-metadata">

**Author:** ![nazar-pc](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/nazar-pc/32/369_2.png) [@nazar-pc](https://mediasoup.discourse.group/u/nazar-pc)\
**Post date:** [July 23, 2021, 9:48am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/6 "2021-07-23T09:48:32Z")

</div>

Try `ffprobe` and see what it says about the file.

---

<div class="post-metadata">

**Author:** ![ibc](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/ibc/32/1540_2.png) [@ibc](https://mediasoup.discourse.group/u/ibc)\
**Post date:** [July 23, 2021, 10:09am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/7 "2021-07-23T10:09:56Z")

</div>

If you take a look to test folder in h246-profile-level-id library (created by us and used in mediasoup) you may figure it out. Otherwise you have to read the H264 RFC spec.

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 23, 2021, 10:17am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/8 "2021-07-23T10:17:35Z")

</div>

I tried the value ffmpeg geneated in sdp file & the value from chrome://webrtc-internals page: 640028、640032、640015, none works

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 23, 2021, 10:18am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/9 "2021-07-23T10:18:22Z")

</div>

% ffprobe ./test.mp4  
…

```auto
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from './test.mp4':
  Metadata:
    major_brand : isom
    minor_version : 512
    compatible_brands: isomiso2avc1mp41
    encoder : Lavf58.76.100
  Duration: 00:01:31.61, start: 0.000000, bitrate: 1295 kb/s
  Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 640x360 [SAR 1:1 DAR 16:9], 1158 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
    Metadata:
      handler_name : Core Media Video
      vendor_id : [0][0][0][0]
  Stream #0:1(zho): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      handler_name : Core Media Audio
      vendor_id : [0][0][0][0]

```

---

<div class="post-metadata">

**Author:** ![nazar-pc](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/nazar-pc/32/369_2.png) [@nazar-pc](https://mediasoup.discourse.group/u/nazar-pc)\
**Post date:** [July 23, 2021, 10:24am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/10 "2021-07-23T10:24:05Z")

</div>

H264 High profile will not work in many browsers when sent over WebRTC (it will using `<video>` tag though).

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 23, 2021, 10:37am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/11 "2021-07-23T10:37:17Z")

</div>

I have converted the test.mp4 to constrained basic profile, but still not works:

```auto
% ffmpeg -i ./test_640p.mp4 -vcodec libx264 -acodec aac -b:a 256k -profile:v baseline test_640p_basic.mp4

```

The symptom: chrome page cannot receive RTP video stream data.

mediasoup server side log:

```auto
  mediasoup:Channel request() [method:transport.consume, id:6] +5ms
  mediasoup:Channel [pid:87109] RTC::SimpleConsumer::CreateRtpStream() | [ssrc:365044946, payloadType:96] +1ms
  mediasoup:Channel [pid:87109] RTC::SimpleConsumer::CreateRtpStream() | FIR supported +0ms
  mediasoup:Channel [pid:87109] RTC::SimpleConsumer::CreateRtpStream() | NACK supported +0ms
  mediasoup:Channel [pid:87109] RTC::SimpleConsumer::CreateRtpStream() | PLI supported +0ms
  mediasoup:Channel request succeeded [method:transport.consume, id:6] +1ms
  mediasoup:Consumer constructor() +0ms
  mediasoup:WebRtcTransport connect() +18ms
  mediasoup:Channel request() [method:transport.connect, id:7] +10ms
  mediasoup:Channel request succeeded [method:transport.connect, id:7] +1ms
  mediasoup:Channel [pid:87109] RTC::IceServer::HandleTuple() | transition from state 'new' to 'connected' +8ms
  mediasoup:Channel [pid:87109] RTC::WebRtcTransport::OnIceServerSelectedTuple() | ICE selected tuple +0ms
  mediasoup:Channel [pid:87109] RTC::WebRtcTransport::OnIceServerConnected() | ICE connected +0ms
  mediasoup:Channel [pid:87109] RTC::WebRtcTransport::MayRunDtlsTransport() | running DTLS transport in local role 'server' +1ms
  mediasoup:Channel [pid:87109] RTC::WebRtcTransport::OnDtlsTransportConnecting() | DTLS connecting +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::Run() | running [role:server] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | DTLS handshake start +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'before SSL initialization'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | role: server, waiting:'before SSL initialization'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'before SSL initialization'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS read client hello'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS write server hello'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS write certificate'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS write key exchange'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS write certificate request'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS write server done'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | role: server, waiting:'SSLv3/TLS write server done'] +1ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS write server done'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS read client certificate'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS read client key exchange'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS read certificate verify'] +1ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS read change cipher spec'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS read finished'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS write change cipher spec'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | [role:server, action:'SSLv3/TLS write finished'] +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::OnSslInfo() | DTLS handshake done +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::CheckRemoteFingerprint() | valid remote fingerprint +0ms
  mediasoup:Channel [pid:87109] RTC::DtlsTransport::GetNegotiatedSrtpCryptoSuite() | chosen SRTP crypto suite: SRTP_AEAD_AES_256_GCM +0ms
  mediasoup:Channel [pid:87109] RTC::WebRtcTransport::OnDtlsTransportConnected() | DTLS connected +0ms
  mediasoup:Consumer resume() +33ms
  mediasoup:Channel request() [method:consumer.resume, id:8] +11ms
  mediasoup:Channel request succeeded [method:consumer.resume, id:8] +1ms
  mediasoup:Channel [pid:87109] RTC::Transport::HandleRtcpPacket() | PLI received, requesting key frame for Consumer [sender ssrc:1, media ssrc:365044946] +1ms
  mediasoup:Channel [pid:87109] RTC::IceServer::HandleTuple() | transition from state 'connected' to 'completed' +33ms
  mediasoup:Channel [pid:87109] RTC::WebRtcTransport::OnIceServerCompleted() | ICE completed +0ms

```

---

<div class="post-metadata">

**Author:** ![nazar-pc](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/nazar-pc/32/369_2.png) [@nazar-pc](https://mediasoup.discourse.group/u/nazar-pc)\
**Post date:** [July 23, 2021, 10:43am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/12 "2021-07-23T10:43:26Z")

</div>

What do you mean by “not works”? I don’t see any errors server-side, check browser logs and dev tools, see if it found decoder for this video.

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 23, 2021, 10:52am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/13 "2021-07-23T10:52:53Z")

</div>

![20210723185115](https://global.discourse-cdn.com/free1/uploads/mediasoup/original/2X/5/569854b88183d4b1bcee83bf397c582a137783f2.jpeg)

from chrome://webrtc-internals page, i can see 2 rtc inbound rtp video stream, no data received displayed here, normally there is H264 encoding info here

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 23, 2021, 10:57am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/14 "2021-07-23T10:57:19Z")

</div>

normally if i use chrome page client to publish, mediasoup server log:

```auto
  mediasoup:Producer constructor() +0ms
  mediasoup:Channel [pid:87330] RTC::IceServer::HandleTuple() | transition from state 'connected' to 'completed' +41ms
  mediasoup:Channel [pid:87330] RTC::WebRtcTransport::OnIceServerCompleted() | ICE completed +0ms
  mediasoup:Channel [pid:87330] RTC::Producer::GetRtpStream() | ignoring RTX packet for not yet created RtpStream (ssrc lookup) +137ms
  mediasoup:WARN:Channel [pid:87330] RTC::Producer::ReceiveRtpPacket() | no stream found for received packet [ssrc:3741614746] +0ms
  mediasoup:Channel [pid:87330] RTC::Producer::CreateRtpStream() | [encodingIdx:0, ssrc:2379022241, rid:, payloadType:123] +0ms
  mediasoup:Channel [pid:87330] RTC::Producer::CreateRtpStream() | FIR supported +0ms
  mediasoup:Channel [pid:87330] RTC::Producer::CreateRtpStream() | NACK supported +0ms
  mediasoup:Channel [pid:87330] RTC::Producer::CreateRtpStream() | PLI supported +0ms
  mediasoup:Channel [pid:87330] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:2379022241, seq:4653] +0ms

```

and then open subscribe page:

```auto
  mediasoup:Channel [pid:87330] RTC::Transport::HandleRtcpPacket() | PLI received, requesting key frame for Consumer [sender ssrc:1, media ssrc:399602566] +0ms
  mediasoup:Channel [pid:87330] RTC::IceServer::HandleTuple() | transition from state 'connected' to 'completed' +52ms
  mediasoup:Channel [pid:87330] RTC::WebRtcTransport::OnIceServerCompleted() | ICE completed +0ms
  mediasoup:Channel [pid:87330] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:2379022241, seq:22431] +1ms
  mediasoup:Channel [pid:87330] RTC::SimpleConsumer::SendRtpPacket() | sync key frame received +0ms
  mediasoup:Channel [pid:87330] RTC::SimpleConsumer::SendRtpPacket() | sending sync packet [ssrc:399602566, seq:1, ts:503037965] from original [seq:22431] +0ms

```

The difference is, when using ffmpeg/gstream to do RTP stream push, seems no “key frame received” log?

---

<div class="post-metadata">

**Author:** ![nazar-pc](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/nazar-pc/32/369_2.png) [@nazar-pc](https://mediasoup.discourse.group/u/nazar-pc)\
**Post date:** [July 23, 2021, 11:23am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/15 "2021-07-23T11:23:08Z")

</div>

What is GOP/key frame interval in that video? It might be expected behavior if you have key frame interval of 8 seconds to have no video for around 8 seconds.

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 23, 2021, 3:51pm UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/16 "2021-07-23T15:51:47Z")

</div>

i wait about ~20s but there is no response in subscribe page.  
i guess it’s not `GOP` problem.  
something error happened, causing the RTP streaming stuck

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 26, 2021, 6:06am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/17 "2021-07-26T06:06:49Z")

</div>

according to [使用 WebRTC 的 GPU 加速流式传输 &nbsp;|&nbsp; 云架构中心 &nbsp;|&nbsp; Google Cloud](https://cloud.google.com/architecture/gpu-accelerated-streaming-using-webrtc?hl=zh-cn)  
i changed rtp config to profile-level-id=42001f，payloadType=102  
but still not working

---

<div class="post-metadata">

**Author:** ![nazar-pc](https://yyz2.discourse-cdn.com/free1/user_avatar/mediasoup.discourse.group/nazar-pc/32/369_2.png) [@nazar-pc](https://mediasoup.discourse.group/u/nazar-pc)\
**Post date:** [July 26, 2021, 7:17am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/18 "2021-07-26T07:17:46Z")

</div>

There must be something else. Maybe port or ssrc is incorrect in ffmpeg command.

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 26, 2021, 10:39am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/19 "2021-07-26T10:39:28Z")

</div>

i checked the `browser page ==> browser page` WebRTC streaming case, in chrome://webrtc-internals , i could see that the outbound video stream and the inbound one have different SSRC, also, why there is 2 inbound video streams?

initially, when i used the mediasoup’s official ffmpeg rtp push case(H264 file converted to VP8 encoding, everything works fine.) but why when i tried to directly use H264 video data, the streaming proc is blocked by something unknown?

---

<div class="post-metadata">

**Author:** ![chenzx](https://avatars.discourse-cdn.com/v4/letter/c/ee59a6/32.png) [@chenzx](https://mediasoup.discourse.group/u/chenzx)\
**Post date:** [July 28, 2021, 6:23am UTC](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072/20 "2021-07-28T06:23:34Z")

</div>

When i modify the ffmpeg RTP push stream command string by changing “libvpx” to “libx264”, H264 streaming is OK, except for “profile-level-id” is re-negotiated from “4d0032” in setting to “4d0015”.

I can see the server log says:

```auto
 mediasoup:Channel [pid:12857] RTC::Producer::ReceiveRtpPacket() | key frame received [ssrc:22222222, seq:8122]

```

But the browser page consumer/player side is not playing video fluent, looks like “jumping from keyframe to next keyframe”? wierd.

But my mp4 file is already H264 encoded, why need re-encoding to let everything OK?

I’ll retry gstreamer case.

[Next page](https://mediasoup.discourse.group/t/does-mediasoup-support-h264-streaming-to-chrome-browser/3072.md?page=2)