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atyenoria / janus-webrtc-gateway-docker

Perfect Docker Image for Media Streaming Expert User ( https://github.com/meetecho/janus-gateway )
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webRTC client -> RTP_FORWARD -> Janus -> FFMPEG -> Facebook RTMP #2

Closed vmintam closed 1 year ago

vmintam commented 7 years ago

Dear bro,

I tried build and use some code in your repository. My goal want to stream my mobile to facebook rtmp.

my rtp forward: "request": "rtp_forward", publisher_id: myid, room: room_dynamic_id, "audio_port": 10033, "audiopt": 111, "video_port": 10038, "videopt": 100, "host": "xxx.xxx.xxx.xxx", "secret": "adminpwd"

everything is fine.

next step, i save my sdp file to janus.sdp like : v=0 o=- 0 0 IN IP4 127.0.0.1 s=RTP Video c=IN IP4 127.0.0.1 t=0 0 a=tool:libavformat 56.15.102 m=audio 10033 RTP/AVP 111 a=rtpmap:111 OPUS/48000/2 m=video 10038 RTP/AVP 100 a=rtpmap:100 VP8/90000 a=fmtp:100 packetization-mode=1

but when i stream to facebook rtmp, i can not see anything:

/root/bin/ffmpeg -analyzeduration 300M -probesize 300M -i janus.sdp -c:v copy -c:a aac -ar 16k -ac 1 -preset ultrafast -tune zerolatency -f flv "rtmp://rtmp-api.facebook.com:80/rtmp/1550595998293270?ds=1&s_l=1&a=ATjqEJKy6ocBD-IF"

pls help me my problem if you free.

thanks!

atyenoria commented 7 years ago

Could you provide the error log from ffmpeg cmd ?

/root/bin/ffmpeg -analyzeduration 300M -probesize 300M -i janus.sdp -c:v copy -c:a aac -ar 16k -ac 1 -preset ultrafast -tune zerolatency -f flv "rtmp://rtmp-api.facebook.com:80/rtmp/1550595998293270?ds=1&s_l=1&a=ATjqEJKy6ocBD-IF"
vmintam commented 7 years ago

hi bro,

here is log : 

ffmpeg -analyzeduration 300M -probesize 300M -i /tmp/janus.sdp -c:v copy -c:a aac -ar 16k -ac 1 -preset ultrafast -tune zerolatency -f flv "rtmp://rtmp-api.facebook.com:80/rtmp/1551408328212037?ds=1&s_l=1&a=ATjRyCQo4Mkq94uX"
ffmpeg version 2.6.8 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --enable-libfaac --enable-nonfree --enable-libfdk-aac --enable-nonfree --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  libavutil      54. 20.100 / 54. 20.100
  libavcodec     56. 26.100 / 56. 26.100
  libavformat    56. 25.101 / 56. 25.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 11.102 /  5. 11.102
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100

[opus @ 0x17bae60] RTP: missed 1 packets
[opus @ 0x17bae60] RTP: dropping old packet received too late
[opus @ 0x17bae60] RTP: missed 2 packets
[opus @ 0x17bae60] RTP: dropping old packet received too late
    Last message repeated 1 times
[sdp @ 0x17b46a0] Could not find codec parameters for stream 1 (Video: vp8, yuv420p): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, sdp, from '/tmp/janus.sdp':
  Metadata:
    title           : RTP Video
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
    Stream #0:1: Video: vp8, yuv420p, 90k tbr, 90k tbn, 90k tbc
Codec AVOption preset (Set the encoding preset (cf. x264 --fullhelp)) specified for output file #0 (rtmp://rtmp-api.facebook.com:80/rtmp/1551408328212037?ds=1&s_l=1&a=ATjRyCQo4Mkq94uX) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
Codec AVOption tune (Tune the encoding params (cf. x264 --fullhelp)) specified for output file #0 (rtmp://rtmp-api.facebook.com:80/rtmp/1551408328212037?ds=1&s_l=1&a=ATjRyCQo4Mkq94uX) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
[aac @ 0x17b7740] The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it.
[aac @ 0x17b7740] Alternatively use the non experimental encoder 'libfaac'.

ffmeg

thanks!

atyenoria commented 7 years ago

you should use h.264 by chaning janus conf file. Now, you use vp8 as a webrtc stream. If you don't want to use vp8, change the -c:v copy to -c:v h264 of ffmpeg cmd.

vmintam commented 7 years ago

hi bro,

i have changed ffmpeg command to : 

ffmpeg -analyzeduration 300M -probesize 300M -i /tmp/janus.sdp -c:v h264 -c:a aac -ar 16k -ac 1 -preset ultrafast -tune zerolatency -f flv "rtmp://rtmp-api.facebook.com:80/rtmp/1551434488209421?ds=1&s_l=1&a=ATg7sl_U3gqQrSCA"

but still error: 

    Last message repeated 1 times
[opus @ 0x1c63e60] RTP: missed 2 packets
[opus @ 0x1c63e60] RTP: dropping old packet received too late
[opus @ 0x1c63e60] RTP: missed 1 packets
    Last message repeated 2 times
[opus @ 0x1c63e60] RTP: missed 2 packets
[opus @ 0x1c63e60] RTP: missed 5 packets
[opus @ 0x1c63e60] RTP: missed 7 packets
[opus @ 0x1c63e60] RTP: dropping old packet received too late
    Last message repeated 3 times
[opus @ 0x1c63e60] RTP: missed 1 packets
[opus @ 0x1c63e60] RTP: dropping old packet received too late
[opus @ 0x1c63e60] RTP: missed 1 packets
[opus @ 0x1c63e60] RTP: dropping old packet received too late
[opus @ 0x1c63e60] RTP: missed 1 packets
[opus @ 0x1c63e60] RTP: dropping old packet received too late
[sdp @ 0x1c5d6a0] Could not find codec parameters for stream 1 (Video: vp8, yuv420p): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, sdp, from '/tmp/janus.sdp':
  Metadata:
    title           : RTP Video
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
    Stream #0:1: Video: vp8, yuv420p, 90k tbr, 90k tbn, 90k tbc
Codec AVOption preset (Set the encoding preset (cf. x264 --fullhelp)) specified for output file #0 (rtmp://rtmp-api.facebook.com:80/rtmp/1551434488209421?ds=1&s_l=1&a=ATg7sl_U3gqQrSCA) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
Codec AVOption tune (Tune the encoding params (cf. x264 --fullhelp)) specified for output file #0 (rtmp://rtmp-api.facebook.com:80/rtmp/1551434488209421?ds=1&s_l=1&a=ATg7sl_U3gqQrSCA) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.

pls suggest me something!

thanks!

atyenoria commented 7 years ago

It seems to be easy to set h264 by changing janus conf. I'm not familiar with this errors....

vmintam commented 7 years ago

ok, i will try janus room plugin config. change to h264.

thanks!

4i-helpynet commented 7 years ago

@vmintam In your output stream you see any video stream? Check it, probably after the rtp_forward the ffmpeg not recognise the source VP8 codec. Audio still okay, but the result is black (or no) video.

@atyenoria At the current Janus version the codec isn't switch to anyting even you setup in the videoroomtest.cfg

atyenoria commented 7 years ago

@4i-helpynet If so, you should create the new issue for that in Janus repo. We can use VP9-SVC in Janus. So, the only h.264 is not good.

4i-helpynet commented 7 years ago

Dear @atyenoria , as I try out now, I cannot change the codec - or VP8 started, even if I declare the VP9 codec. Can you send me your config file?

atyenoria commented 7 years ago

@4i-helpynet No need for that. I just changed the janus video conf line in official repo. There is no special conf.

4i-helpynet commented 7 years ago

@atyenoria well, I try out again but at the room creation still respond me as VP8 codec...

Yep, my conf in videoroomtest.cfg is: [1234] description = Demo Room secret = adminpwd publishers = 6 bitrate = 128000 fir_freq = 1 ;audiocodec = opus videocodec = vp9 record = false ;rec_dir = /path/to/recordings-folder

And I receive: {janus: "event", session_id: 67248937191837, sender: 8004835962635332, transaction: "ZmFwSH5LUfST",…} janus:"event" jsep:{type: "answer",…} plugindata:{plugin: "janus.plugin.videoroom",…} data:{videoroom: "event", room: 1234, configured: "ok", audio_codec: "opus", video_codec: "vp8"} audio_codec:"opus "configured:"ok" room:1234 video_codec:"vp8" videoroom:"event" plugin:"janus.plugin.videoroom" sender:8004835962635332 session_id:67248937191837 transaction:"ZmFwSH5LUfST"

atyenoria commented 7 years ago

@4i-helpynet Can you change the codec to h.264 instead of vp9?

atyenoria commented 7 years ago 
[1234]
description = Demo Room
publishers = 6
bitrate = 1000000
fir_freq = 10
;audiocodec = opus
videocodec = h264
record = true
rec_dir = /tmp/record-test
atyenoria commented 7 years ago

or Probably, you always load the wrong config file which is the default file. So, I think that you can't change any parameters. right? Check the loaded file.

4i-helpynet commented 7 years ago

I try out. I change the janus.cfg (turn api secret on/off): Working: I change the janus.plugin.videoroom.cfg (not the sample), and it isn't change

Maybe that codec line is affect only to the stream recording, not the browser's?

atyenoria commented 7 years ago

no. That affects the browser and recording.

4i-helpynet commented 7 years ago

if you change the codec, in the browser at the janus JSON change to the selected codec too? (and what version of Janus do you use?)

atyenoria commented 7 years ago

yes.

* 948f3cb - (3 months ago) Give ICE 5 seconds before considering failed a definitive state ― Lorenzo Miniero (origin/icefailed-timer)
4i-helpynet commented 7 years ago

That was an error of Janus - fixed now.

atyenoria commented 7 years ago

Okay

ivanovaleksey commented 6 years ago

Hello guys, I have the same issue thought I try to stream not to RTMP but to plain file. Unfortunately, I got only audio stream in output file (and black screen for video).

I created the following SDP:

v=0
o=- 0 0 IN IP4 127.0.0.1
s=RTP Video
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 56.15.102
m=audio 10033 RTP/AVP 111
a=rtpmap:111 OPUS/48000/2
m=video 10038 RTP/AVP 100
a=rtpmap:100 VP8/90000

and run command:

ffmpeg \
  -protocol_whitelist file,udp,rtp  -i sdp.test \
  -c:v libx264 -c:a aac \
  -y test.mp4

Ffmpeg output is:

ffmpeg version 3.4-4~16.04.york0 Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.5) 20160609
  configuration: --prefix=/usr --extra-version='4~16.04.york0' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
[sdp @ 0x5645afc2bec0] Could not find codec parameters for stream 1 (Video: vp8, yuv420p): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, sdp, from 'sdp.test':
  Metadata:
    title           : RTP Video
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
    Stream #0:1: Video: vp8, yuv420p, 90k tbr, 90k tbn, 90k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> aac (native))
Press [q] to stop, [?] for help
Output #0, mp4, to 'test.mp4':
  Metadata:
    title           : RTP Video
    encoder         : Lavf57.83.100
    Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc57.107.100 aac

So, ffmpeg doesn't see output video stream. I have googled all around but I can't find an answer.

I try to add -analyzeduration 300M -probesize 300M but it doesn't work. Ffmpeg just hangs very long and then prints the same output.

  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
[sdp @ 0x55cc0ca87f00] Could not find codec parameters for stream 1 (Video: vp8, yuv420p): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, sdp, from 'sdp.test':
  Metadata:
    title           : RTP Video
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
    Stream #0:1: Video: vp8, yuv420p, 90k tbr, 90k tbn, 90k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> aac (native))
Press [q] to stop, [?] for help
Output #0, mp4, to 'test.mp4':
  Metadata:
    title           : RTP Video
    encoder         : Lavf57.83.100
    Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc57.107.100 aac
[sdp @ 0x55cc0ca87f00] max delay reached. need to consume packet3x
[sdp @ 0x55cc0ca87f00] RTP: missed 283 packets
size=    5247kB time=00:05:37.52 bitrate= 127.4kbits/s speed=9.01x

I use Janus Videoroom plugin. After joining a room I send rtp_forward request via console:

var message = { "request": "rtp_forward", publisher_id: publisherId, room: 1234, "audio_port": 10033, "audiopt": 111, "video_port": 10038, "videopt": 100, "host": "127.0.0.1", "secret": "adminpwd" };
sfutest.send({"message": message});

Could you please help me to figure out the problem? Thanks.

atyenoria commented 6 years ago

@ivanovaleksey I'm not sure of how to solve this? Did you set the FIR to 1 by janus config for test?

BTW, you could build the ffmpeg with --enable-libmp3lame How did you achieve this?! I can't build that in my docker build....

ivanovaleksey commented 6 years ago

@atyenoria thank you for advices,

Yes, I set FIR to 1. Seems like the problem was with my SDP. Actual video stream has RTP payload type 96 but SDP was looking was 100. Very stupid of mine :|

By the way, can the issue be closed?

saroar commented 5 years ago

@4i-helpynet hi can you help me a little bit please here is my skype saroarkhandoker1

AlexanderKozhevin commented 5 years ago

Sorry, where can I get "publisher_id" ? Im getting the same error "no publisher id".

@atyenoria

@vmintam

Screenshot 2019-06-23 at 23 35 13

dvision1979 commented 4 years ago

That was an error of Janus - fixed now.

@4i-helpynet Did you manage to use the h264 codec with janus videoroom plugin for rtp_forward. I am trying all kinds of variations for SDPs, but with no success. Could you please help me with some suggestions? Thank you

4i-helpynet commented 4 years ago

@danionescu2007 You have to check the videopt & audiopt and ports to the same as you see in the console after the successful publishing and rtp_forward request. My method was : I run FFMPEG in console, and check if the encoding started with the given SDP infos. Finally I able to set. VP8: m=audio [PORT] RTP/AVP 111 a=rtpmap:111 OPUS/48000/2 m=video [PORT] RTP/AVP 96 a=rtpmap:96 VP8/90000

h264: m=audio [PORT] RTP/AVP 111 a=rtpmap:111 OPUS/48000/2 m=video [PORT] RTP/AVP 100 a=rtpmap:100 H264/90000

dvision1979 commented 4 years ago

thank you very much.

On Thu, Sep 10, 2020 at 11:35 AM 4i-helpynet notifications@github.com wrote:

@danionescu2007 https://github.com/danionescu2007 You have to check the videopt & audiopt and ports to the same as you see in the console after the successful publishing and rtp_forward request. My method was : I run FFMPEG in console, and check if the encoding started with the given SDP infos. Finally I able to set. VP8: m=audio [PORT] RTP/AVP 111 a=rtpmap:111 OPUS/48000/2 m=video [PORT] RTP/AVP 96 a=rtpmap:96 VP8/90000

h264: m=audio [PORT] RTP/AVP 111 a=rtpmap:111 OPUS/48000/2 m=video [PORT] RTP/AVP 100 a=rtpmap:100 H264/90000

— You are receiving this because you were mentioned. Reply to this email directly, view it on GitHub https://github.com/atyenoria/janus-webrtc-gateway-docker/issues/2#issuecomment-690083187, or unsubscribe https://github.com/notifications/unsubscribe-auth/ABDYVYRL6F7T3ANJTABATNDSFCFWPANCNFSM4DO3AHUA .

--

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hohno-panopto commented 3 years ago

@danionescu2007 By any chance, can you share a) the video room definition in janus.plugin.videoroom.jcfg, b) entire sdp file which is passed to ffmpeg, c) entire command line passed to ffmpeg, and d) the command which is sent to the video room plug-in to start RTP forwarder?

I tried various things mentioned in this conversation with the latest Janus, but ffmpeg never recognizes incoming H.264 and relay does not work for me. I want to replicate your working environment all if possible. I confirmed audio side works by removing video. I also tried VP8 (with ffmpeg transcode) and RTMP relay scenario works.

4i-helpynet commented 3 years ago

I suggest to try out a working method with OPUS and VP8 codec.

After that the h264 method will be easier to setup - (I find more difficult to set in h264, and the encoding is more seamless in VP8 - so, if you doesn't have to stream specially from iOS safari, you should use VP8-VP9.

Üdvözlettel Munkatársaink egyeztetés után elérhetőek virtuális irodánkban: https://4i.hu/virtualisiroda/ https://4i.hu/virtualisiroda/

This message and any attachment are confidential and are legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. Please note that any dissemination, distribution, copying or use of or reliance upon the information contained in and transmitted with this e-mail by or to anyone other than the recipient designated above by the sender is unauthorised and strictly prohibited.

Hiroshi Ohno notifications@github.com ezt írta (időpont: 2021. febr. 7., V, 18:16):

@danionescu2007 https://github.com/danionescu2007 By any chance, can you share a) the video room definition in janus.plugin.videoroom.jcfg, b) entire sdp file which is passed to ffmpeg, c) entire command line passed to ffmpeg, and d) the command which is sent to the video room plug-in to start RTP forwarder?

I tried various things mentioned in this conversation with the latest Janus, but ffmpeg never recognizes incoming H.264 and relay does not work for me. I want to replicate your working environment all if possible. I confirmed audio side works by removing video. I also tried VP8 (with ffmpeg transcode) and RTMP relay scenario works.

— You are receiving this because you were mentioned. Reply to this email directly, view it on GitHub https://github.com/atyenoria/janus-webrtc-gateway-docker/issues/2#issuecomment-774711777, or unsubscribe https://github.com/notifications/unsubscribe-auth/AE3FI2XYT5UFKOIUPOEHKYDS53DFLANCNFSM4DO3AHUA .

hohno-panopto commented 3 years ago

Check the correct port and rtp map settings both in janus server, both the response of server, and the FFMPEG .sdp part After that the h264 method will be easier to setup - (I find more difficult to set in h264, and the encoding is more seamless in VP8 - so, if you doesn't have to stream specially from iOS safari, you should use VP8-VP9.

Thank you @4i-helpynet. (Actually, I meant to ask you, not Dan)My end goal is to pull out the media outside of webRTC world in scale, and H.264 is strongly desired to avoid transcoding on the server.I will try once more with your suggestion.

4i-helpynet commented 3 years ago

Hi

Please consider that the trascoding will be unavoidable most of the cases - in other words, a stream without transcodig is 0.1-0.2% of user cases. If you want multiple version of stream depended of bandwith you have to transcode anyway.

Üdvözlettel Munkatársaink egyeztetés után elérhetőek virtuális irodánkban: https://4i.hu/virtualisiroda/ https://4i.hu/virtualisiroda/

This message and any attachment are confidential and are legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. Please note that any dissemination, distribution, copying or use of or reliance upon the information contained in and transmitted with this e-mail by or to anyone other than the recipient designated above by the sender is unauthorised and strictly prohibited.

Hiroshi Ohno notifications@github.com ezt írta (időpont: 2021. febr. 9., K, 7:47):

Check the correct port and rtp map settings both in janus server, both the response of server, and the FFMPEG .sdp part After that the h264 method will be easier to setup - (I find more difficult to set in h264, and the encoding is more seamless in VP8 - so, if you doesn't have to stream specially from iOS safari, you should use VP8-VP9.

Thank you @4i-helpynet https://github.com/4i-helpynet. (Actually, I meant to ask you, not Dan)My end goal is to pull out the media outside of webRTC world in scale, and H.264 is strongly desired to avoid transcoding on the server.I will try once more with your suggestion.

— You are receiving this because you were mentioned. Reply to this email directly, view it on GitHub https://github.com/atyenoria/janus-webrtc-gateway-docker/issues/2#issuecomment-775712871, or unsubscribe https://github.com/notifications/unsubscribe-auth/AE3FI2WIMUL6WURFV46OP5LS6DLA7ANCNFSM4DO3AHUA .