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• Streaming RTP with ffmpeg and node.js to voip phone

  5 juillet 2023, par Nik Hendricks

  I am trying to implement SIP in node.js. Here is the library i am working on

  


  Upon receiving an invite request such as

  


  
Received INVITE
INVITE sip:201@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.39:5062;branch=z9hG4bK1534941205
From: "Nik" <sip:nik@192.168.1.2>;tag=564148403
To: <sip:201@192.168.1.2>
Call-ID: 2068254636@192.168.1.39
CSeq: 2 INVITE
Contact: <sip:nik@192.168.1.39:5062>
Authorization: Digest username="Nik", realm="NRegistrar", nonce="1234abcd", uri="sip:201@192.168.1.2:5060", response="7fba16dafe3d60c270b774bd5bba524c", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.71.0.120
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 306

v=0
o=- 20083 20083 IN IP4 192.168.1.39
s=SDP data
c=IN IP4 192.168.1.39
t=0 0
m=audio 11782 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv



  


  I can then parse the SDP into an object like this

  


   
{
    "session":{
        "version":"0",
        "origin":"- 20084 20084 IN IP4 192.168.1.39",
        "sessionName":"SDP data"
    },
    "media":[
        {
            "media":"audio",
            "port":11784,
            "protocol":"RTP/AVP",
            "format":"0",
            "attributes":[
                "rtpmap:0 PCMU/8000",
                "rtpmap:8 PCMA/8000",
                "rtpmap:18 G729/8000",
                "fmtp:18 annexb=no",
                "rtpmap:9 G722/8000",
                "fmtp:101 0-15",
                "rtpmap:101 telephone-event/8000",
                "ptime:20",
                "sendrecv"
            ]
        }
    ]
}


  


  After sending the 100 and 180 responses with my library i attempt to start a RTP stream with ffmpeg

  


  var port = SDPParser.parse(res.message.body).media[0].port
var s = new STREAMER('output.wav', '192.168.1.39', port)


  


  with the following STREAMER class

  


  class Streamer{
    constructor(inputFilePath, rtpAddress, rtpPort){
        this.inputFilePath = 'output.wav';
        this.rtpAddress = rtpAddress;
        this.rtpPort = rtpPort;
    }

    start(){
        return new Promise((resolve) => {
            const ffmpegCommand = `ffmpeg -re -i ${this.inputFilePath} -ar 8000 -f mulaw -f rtp rtp://${this.rtpAddress}:${this.rtpPort}`;
            const ffmpegProcess = spawn(ffmpegCommand, { shell: true });
    
            ffmpegProcess.stdout.on('data', (data) => {
                data = data.toString()
                //replace all instances of 127.0.0.1 with our local ip address
                data = data.replace(new RegExp('127.0.0.1', 'g'), '192.168.1.3');

                resolve(data.toString())
            });
    
            ffmpegProcess.stderr.on('data', (data) => {
              // Handle stderr data if required
              console.log(data.toString())
            });
    
            ffmpegProcess.on('close', (code) => {
              // Handle process close event if required
              console.log('close')
              console.log(code.toString())
            });
    
            ffmpegProcess.on('error', (error) => {
              // Handle process error event if required
              console.log(error.toString())
            });
        })
    }
     
}


  


  the start() function resolves with the SDP that ffmpeg generates. I am starting to think that ffmpeg cant generate proper SDP for voip calls.

  


  so when i create 200 response with the following sdp

  


  v=0
o=- 0 0 IN IP4 192.168.1.3
s=Impact Moderato
c=IN IP4 192.168.1.39
t=0 0
a=tool:libavformat 58.29.100
m=audio 12123 RTP/AVP 97
b=AS:128
a=rtpmap:97 PCMU/8000/2


  


  the other line never picks up. from my understanding the first invite from the caller will provide SDP that will tell me where to send the RTP stream too and the correct codecs and everything. I know that currently, my wav file is PCMU and i can listen to it with ffplay and the provided sdp. what is required to make the other line pickup specifically a Yealink t42g

  


  my full attempt looks like this

  


  Client.on('INVITE', (res) => {
    console.log("Received INVITE")
    var d = Client.Dialog(res).then(dialog => {
        dialog.send(res.CreateResponse(100))
        dialog.send(res.CreateResponse(180))
        var port = SDPParser.parse(res.message.body).media[0].port

        var s = new STREAMER('output.wav', '192.168.1.39', port)
        s.start().then(sdp => {
            console.log(sdp.split('SDP:')[1])
            var ok = res.CreateResponse(200)
            ok.body = sdp.split('SDP:')[1]
            dialog.send(ok)
        })

        dialog.on('BYE', (res) => {
            console.log("BYE")
            dialog.send(res.CreateResponse(200))
            dialog.kill()
        })
    })
})


  


  I have provided a link to my library at the top of this message. My current problem is in the examples/Client folder.

  


  I'm not sure what could be going wrong here. Maybe i'm not using the right format or codec for the VOIP phone i dont see whats wrong with the SDP. especially if i can listen to SDP generated by ffmpeg if i stream RTP back to the same computer i use ffplay on. Any help is greatly appreciated.

  


  Update

  


  As i test i decided to send the caller back SDP that was generated by a Yealink phone like itself. but with some modifications

  


  v=0
o=- ${this.output_port} ${this.output_port} IN IP4 192.168.1.39
s=SDP data
c=IN IP4 192.168.1.39
t=0 0
m=audio ${this.output_port} RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:1
01 telephone-event/8000
a=ptime:20
a=sendrecv


  


  Finally, the phone that makes the call in the first place will fully answer but still no audio stream. I notice if I change the IP address or port to something wrong the other phone Will hear its own audio instead of just quiet. so this leads me to believe I am headed in the right direction. And maybe the problem lies in not sending the right audio format for what I'm describing.

  


  Additionaly, Whenever using ffmpeg to stream my audio with rtp I notice that it sees the file format as this pcm_alaw, 8000 Hz, mono, s16, 64 kb/s My new SDP describes using both ulaw and alaw but I'm not sure which it is saying it prefers

  


  v=0
o=- ${this.output_port} ${this.output_port} IN IP4 192.168.1.39
s=SDP data
c=IN IP4 192.168.1.39
t=0 0
m=audio ${this.output_port} RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:0
a=sendrecv


  


  I have been able to simply the SDP down to this. This will let the other phone actually pickup and not hear its own audio. it's just a completely dead air stream.

  


• ffmpeg rtmp live stream to Azure Media Services

  18 mars 2015, par Pilskalns

  Somehow I got publishing Azure Media Services to work, except there is huge frame drop (almost every frame) with those settings :

  ffmpeg -i "http://stream01/hls/live/playlist-mid.m3u8" -strict -2 -c:a aac -b:a 192k
  
  -ar 44100 -r 25 -g 60 -keyint_min 60 -b:v 2000k -c:v libx264 -bufsize 2000k
  
  -maxrate 2000k -f flv "rtmp://publishurl.working.is/mystream1"

  + I can’t play URL from Azure.

  When I publish to local nginx-rtmp server, frame drop stays and despite it I can play it. So, I believe that frame drop is ffmpeg related not Azure. Processor is not overloaded, available bandwidth is excellent.

  When I try original settings from this URL (at very bottom), I this get additionaly this ffmpeg error Past duration 0.719994 too large

  Here’s more from output :

  c:\ffmpeg\bin>ffmpeg -i "http://stream01/hls/live/playlist-mid.m3u8" -strict -2 -c:a aac -b:a 192k -ar 44100 -r 25 -g 60 -keyint_min 60 -b:v 2000k -c:
  
  v libx264 -bufsize 1000k -maxrate 2000k -f flv "rtmp://********.channel.mediaservices.windows.net:1935/live/9590f0be/mystream1"
  
  ffmpeg version N-69972-g6c91afe Copyright (c) 2000-2015 the FFmpeg developers
  
    built with gcc 4.9.2 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnu
  
  tls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-
  
  libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-l
  
  ibrtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --
  
  enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --ena
  
  ble-libxvid --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      54. 19.100 / 54. 19.100
  
    libavcodec     56. 25.100 / 56. 25.100
  
    libavformat    56. 23.100 / 56. 23.100
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 11.100 /  5. 11.100
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, hls,applehttp, from 'http://stream01/hls/live/playlist-mid.m3u8':
  
    Duration: N/A, start: 3314.605633, bitrate: N/A
  
    Program 0
  
      Metadata:
  
        variant_bitrate : 1500000
  
      Stream #0:0: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv), 854x480 [SAR 1:1 DAR 427:240], 50 fps, 50 tbr, 90k tbn, 100 tbc
  
      Metadata:
  
        variant_bitrate : 1500000
  
      Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 216 kb/s
  
      Metadata:
  
        variant_bitrate : 1500000
  
  [libx264 @ 0000000000306280] using SAR=1/1
  
  [libx264 @ 0000000000306280] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
  
  [libx264 @ 0000000000306280] profile High, level 3.0
  
  [libx264 @ 0000000000306280] 264 - core 144 r2525 40bb568 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options:
  
   cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 dead
  
  zone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_in
  
  tra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=60 keyint_min=31 scenecut=40 intra_refresh=0 rc_lookahea
  
  d=40 rc=cbr mbtree=1 bitrate=2000 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=2000 vbv_bufsize=1000 nal_hrd=none filler=0 ip_ratio=1.
  
  40 aq=1:1.00
  
  Output #0, flv, to 'rtmp://*******.channel.mediaservices.windows.net:1935/live/37b4e25/mystream1':
  
    Metadata:
  
      encoder         : Lavf56.23.100
  
      Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 854x480 [SAR 1:1 DAR 427:240], q=-1--1, 2000 kb/s, 25 fps, 1k tbn, 25 tbc
  
      Metadata:
  
        variant_bitrate : 1500000
  
        encoder         : Lavc56.25.100 libx264
  
      Stream #0:1: Audio: aac ([10][0][0][0] / 0x000A), 44100 Hz, stereo, fltp, 192 kb/s
  
      Metadata:
  
        variant_bitrate : 1500000
  
        encoder         : Lavc56.25.100 aac
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  
    Stream #0:1 -> #0:1 (aac (native) -> aac (native))
  
  Press [q] to stop, [?] for help
  
  [flv @ 0000000000305820] Failed to update header with correct duration.1.4kbits/s dup=0 drop=10697
  
  [flv @ 0000000000305820] Failed to update header with correct filesize.
  
  frame=10714 fps= 24 q=-1.0 Lsize=  100583kB time=00:07:11.08 bitrate=1911.4kbits/s dup=0 drop=10701
  
  video:90024kB audio:10042kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.516478%
  
  [libx264 @ 0000000000306280] frame I:190   Avg QP:10.58  size: 72438
  
  [libx264 @ 0000000000306280] frame P:3406  Avg QP:14.32  size: 19296
  
  [libx264 @ 0000000000306280] frame B:7118  Avg QP:19.55  size:  1784
  
  [libx264 @ 0000000000306280] consecutive B-frames:  8.8%  4.4% 10.5% 76.3%
  
  [libx264 @ 0000000000306280] mb I  I16..4: 23.0% 20.2% 56.8%
  
  [libx264 @ 0000000000306280] mb P  I16..4:  2.7%  4.5%  5.2%  P16..4: 41.9% 11.6%  8.7%  0.0%  0.0%    skip:25.4%
  
  [libx264 @ 0000000000306280] mb B  I16..4:  0.1%  0.1%  0.2%  B16..8: 26.5%  2.2%  0.5%  direct: 3.2%  skip:67.1%  L0:51.0% L1:38.2% BI:10.8%
  
  [libx264 @ 0000000000306280] 8x8 transform intra:31.3% inter:36.6%
  
  [libx264 @ 0000000000306280] coded y,uvDC,uvAC intra: 80.5% 73.5% 51.1% inter: 16.1% 18.8% 5.8%
  
  [libx264 @ 0000000000306280] i16 v,h,dc,p: 21% 29% 27% 23%
  
  [libx264 @ 0000000000306280] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 32% 19%  4%  4%  4%  5%  4%  7%
  
  [libx264 @ 0000000000306280] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 27% 30% 15%  4%  5%  5%  5%  4%  5%
  
  [libx264 @ 0000000000306280] i8c dc,h,v,p: 38% 30% 24%  8%
  
  [libx264 @ 0000000000306280] Weighted P-Frames: Y:1.8% UV:0.4%
  
  [libx264 @ 0000000000306280] ref P L0: 78.0%  8.5% 10.1%  3.3%  0.0%
  
  [libx264 @ 0000000000306280] ref B L0: 91.3%  7.9%  0.8%
  
  [libx264 @ 0000000000306280] ref B L1: 95.4%  4.6%
  
  [libx264 @ 0000000000306280] kb/s:1716.97
  
  Received signal 2: terminating.

• Panasonc GH5 4K 10 Bit 25p (cannot allocate memory)

  25 novembre 2017, par Sebastian

  I am converting GH5 files with the following script. With my Notebook I get no error and it goes through but at my desktop workstation I get this error. Anybody an idea. I convert 4K 10Bit Panasonic GH5 files to AVID HQX files. No error at the notebook but the desktop pc produces this error message. OS is windows 7 just new installed because of this. Same MeGUI and FFMPEG version and development server.

  enter code here

  D:\Test>for %f in (*.mov) do "C:\Program Files (x86)\MeGUI\tools\ffmpeg\ffmpeg.e
  
  xe" -i "%~f" -c:a pcm_s16le -c:v dnxhd -profile:v dnxhr_hqx "%~nf_test.mov"
  
  
  
  D:\Test>"C:\Program Files (x86)\MeGUI\tools\ffmpeg\ffmpeg.exe" -i "P1011064.MOV"
  
   -c:a pcm_s16le -c:v dnxhd -profile:v dnxhr_hqx "P1011064_test.mov"
  
  ffmpeg version 3.4 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 7.2.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --e
  
  nable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libblur
  
  ay --enable-libfreetype --enable-libmp3lame --enable-libopenjpeg --enable-libopu
  
  s --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --ena
  
  ble-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-lib
  
  x264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-z
  
  lib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-cuda --enable-c
  
  uvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth --enable-l
  
  ibmfx
  
    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
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 005a7860] decoding for stream 0 failed
  
  Guessed Channel Layout for Input Stream #0.1 : stereo
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'P1011064.MOV':
  
    Metadata:
  
      major_brand     : qt
  
      minor_version   : 537986816
  
      compatible_brands: qt  pana
  
      creation_time   : 2017-11-19T16:16:23.000000Z
  
      com.panasonic.Semi-Pro.metadata.xml: <?xml version="1.0" encoding="UTF-8" st
  
  andalone="no" ?>
  
                      : <clipmain xmlns="urn:schemas-Professional-Plug-in:Semi-Pro:ClipMetadata:v1.0">
  
                      :   <clipcontent>
  
                      :     <globalclipid>060A2B340101010501010D2113000000171BA845
  
  E82C2C3470010E90B8D50052</globalclipid>
  
                      :     <duration>84</duration>
  
                      :     <editunit>1/25</editunit>
  
                      :     <essencelist>
  
                      :       <video>
  
                      :         <codec bitrate="150">H264_422_LongGOP</codec>
  
                      :         <activeline>2160</activeline>
  
                      :         <activepixel>3840</activepixel>
  
                      :         <bitdepth>10</bitdepth>
  
                      :         <framerate>25p</framerate>
  
                      :         <timecodetype>NonDrop</timecodetype>
  
                      :         <starttimecode>07:12:44:18</starttimecode>
  
                      :       </video>
  
                      :       <audio>
  
                      :         <channel>2</channel>
  
                      :         <samplingrate>48000</samplingrate>
  
                      :         <bitspersample>16</bitspersample>
  
                      :       </audio>
  
                      :     </essencelist>
  
                      :     <clipmetadata>
  
                      :       <rating>0</rating>
  
                      :       <access>
  
                      :         <creationdate>2017-11-19T16:16:23+02:00
  
                      :         <lastupdatedate>2017-11-19T16:16:23+02:00
  
                      :       </lastupdatedate></creationdate></access>
  
                      :       <device>
  
                      :         <manufacturer>Panasonic</manufacturer>
  
                      :         <modelname>DC-GH5</modelname>
  
                      :       </device>
  
                      :       <shoot>
  
                      :         <startdate>2017-11-19T16:16:23+02:00</startdate>
  
                      :       </shoot>
  
                      :     </clipmetadata>
  
                      :   </clipcontent>
  
                      :   <userarea>
  
                      :     <acquisitionmetadata xmlns="urn:schemas-Professional-P
  
  lug-in:P2:CameraMetadata:v1.2">
  
                      :       <cameraunitmetadata>
  
                      :         <gamma>
  
                      :           <capturegamma>CINELIKE_D</capturegamma>
  
                      :         </gamma>
  
                      :         <gamut>
  
                      :           <capturegamut>BT.709</capturegamut>
  
                      :         </gamut>
  
                      :       </cameraunitmetadata>
  
                      :     </acquisitionmetadata>
  
                      :   </userarea>
  
                      : </clipmain>
  
                      :
  
    Duration: 00:00:03.36, start: 0.000000, bitrate: 174710 kb/s
  
      Stream #0:0(und): Video: h264 (High 4:2:2) (avc1 / 0x31637661), yuv422p10le(
  
  tv, bt709), 3840x2160 [SAR 1:1 DAR 16:9], 147582 kb/s, 25 fps, 25 tbr, 90k tbn,
  
  50 tbc (default)
  
      Metadata:
  
        creation_time   : 2017-11-19T16:16:23.000000Z
  
        timecode        : 07:12:44:18
  
      Stream #0:1(und): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, stereo, s1
  
  6, 1536 kb/s (default)
  
      Metadata:
  
        creation_time   : 2017-11-19T16:16:23.000000Z
  
        timecode        : 07:12:44:18
  
      Stream #0:2(und): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
  
      Metadata:
  
        creation_time   : 2017-11-19T16:16:23.000000Z
  
        timecode        : 07:12:44:18
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> dnxhd (native))
  
    Stream #0:1 -> #0:1 (pcm_s16be (native) -> pcm_s16le (native))
  
  Press [q] to stop, [?] for help
  
  [dnxhd @ 0542fe60] Cannot allocate memory.
  
  [dnxhd @ 04d508a0] ff_frame_thread_encoder_init failed
  
  Error initializing output stream 0:0 -- Error while opening encoder for output s
  
  tream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
  
  Conversion failed!
  
  
  
  D:\Test>pause
  
  Drücken Sie eine beliebige Taste . . .