WebRTC to rtsp/rtmp/hls

[gaurav mishra]

Hi,

I am trying to build a broadcasting application in which there will be two types of viewers 

1) low latency (consuming from WebRTC endpoints )

2)High latency (consuming http video segments from a cdn).

For the second type of viewers i am trying to figure out is there a way to feed the stream from webrtc endpoint to a traditional/older streaming infrastructure  

ie   webRtc --> (something that can get me hls stream) -> cdn 

for the middle part i have tried wowza with rtpEndpoint 

( webRTC-->rtpEndpoint -->wowza ->cdn). But it involves a lot of manual steps and have not been able to figure out a way to do everything programatically. 

Are there any other approach i can take to achieve what i want. even if it involves other transcoding softwares (ffmpeg/gstreamer). basically is there a way to transform webRTC stream into rtsp or rtmp or hls stream?

[Andrey Utkin]

Hi,

We do streaming kurento webrtc endpoint to hls, but with Kurento 5.1. On Kurento side, webrtc inbound endpoint connected to two RTP outbound endpoints - for video and audio streams respectively.

Then create small script wich use GStreamer to listen from RTP endpoints (we use udpsrc) and transcode/mux/create HSL files. Important thing is to create correct SDP files (offers) to setup outbound RTP endpoints on kurento side. In our case they looks like:

"videoSdpOffer":"v=0\no=- 1440770100 1440770100 IN IP4 192.168.56.103\ns=handmade\nc=IN IP4 192.168.56.103\nm=video 50712 RTP/AVP 101\na=rtpmap:101 VP8/90000",

"audioSdpOffer":"v=0\no=- 1440770100 1440770100 IN IP4 192.168.56.103\ns=handmade\nc=IN IP4 192.168.56.103\nm=audio 58740 RTP/AVP 96\na=rtpmap:96 OPUS/48000/2"

воскресенье, 30 августа 2015 г., 11:08:10 UTC+3 пользователь gaurav mishra написал:

[gaurav mishra]

Thanks for the reply. 

I am now able to do the same thing using ffmpeg . ie created two sdp files one for audio and one for video and mux them into a single stream. 

Which brings another question --why do we need to create two sdp files and two rtp endpoint . i tried having both audio and video stream descriptions in same sdp file (also single rtpEndpoint ) but ffmpeg was throwing some errors while processing this sdp file and i was not able to get audio in final output.  

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[Mark Hassman]

Note sure if this helps, but I'm running a similar chain.. webrtc -> kurento -> rtp -> ffmpeg -> hls.

I'm using only a single rtp sdp/endpoint per kurento pipeline. Here's a sample:

 

   var rtp_sdp = "v=0\n" +
    "o=- 3641290734 3641290734 IN IP4 1.1.1.1\n" +
    "s=nginx\n" +
    "c=IN IP4 1.1.1.1\n" +
    "t=0 0\n" +
    "m=audio 60820 RTP/AVP 0\n" +
    "a=rtpmap:0 PCMU/8000\n" +
    "a=recvonly\n" +
    "m=video 59618 RTP/AVP 101\n" +
    "a=rtpmap:101 H264/90000\n" +
    "a=recvonly\n";
(ip's have been changed for privacy, but the idea is the same.. change 1.1.1.1 to your ffmpeg listening ip and pass this type of sdp to kurento's rtp processoffer function)

 

Note: i've found if you setup kurento's rtp endpoint before webrtp is complete, kurento at times sends rtp data to the incorrect udp port - offset by one increment from the sdp... so, i now wait for media state changed event of connected. that seems to have prevented the issue.

 

---

From: kur...@googlegroups.com [mailto:kur...@googlegroups.com] On Behalf Of gaurav mishra
Sent: Monday, August 31, 2015 6:40 AM
To: kur...@googlegroups.com
Subject: Re: [kurento-public] Re: WebRTC to rtsp/rtmp/hls

[Jakeooo]

 webrtc -> rtp -> ffmpeg 

sdp file

root@ubuntu:/tmp# cat test.sdp

v=0

o=SerTest 0 0 IN IP4 0.0.0.0

s=-

c=IN IP4 0.0.0.0

t=0 0

m=video 30010 RTP/AVP 100

a=rtpmap:100 H264/90000

a=recvonly

----------------------------------------------------------------

I have got error like this 

root@ubuntu:/tmp# ffmpeg -i /tmp/test.sdp -g 5 -c copy -f flv rtmp://localhost/hls/test

ffmpeg version N-76045-g97be5d4 Copyright (c) 2000-2015 the FFmpeg developers

  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04)

  configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libdcadec --enable-libfreetype --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvo-aacenc --enable-libvidstab

  libavutil      55.  4.100 / 55.  4.100

  libavcodec     57.  6.100 / 57.  6.100

  libavformat    57.  4.100 / 57.  4.100

  libavdevice    57.  0.100 / 57.  0.100

  libavfilter     6. 11.100 /  6. 11.100

  libavresample   3.  0.  0 /  3.  0.  0

  libswscale      4.  0.100 /  4.  0.100

  libswresample   2.  0.100 /  2.  0.100

  libpostproc    54.  0.100 / 54.  0.100

[h264 @ 0x3bb50c0] non-existing PPS 0 referenced

    Last message repeated 1 times

[h264 @ 0x3bb50c0] decode_slice_header error

[h264 @ 0x3bb50c0] no frame!

[h264 @ 0x3bb50c0] non-existing PPS 0 referenced

    Last message repeated 1 times

Any hint for the issue is appreciated. 

Thanks!

在 2015年9月1日星期二 UTC+8上午7:27:15，Mark Hassman写道：