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Rennes Emotion Map 2010-11
19 octobre 2011, par
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (76)
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Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
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13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...) -
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Au niveau des dépendances logicielles Utilisation des dernières versions de FFMpeg (>= v1.2.1) ; Installation des dépendances pour Smush ; Installation de MediaInfo et FFprobe pour la récupération des métadonnées ; On n’utilise plus ffmpeg2theora ; On n’installe plus flvtool2 au profit de flvtool++ ; On n’installe plus ffmpeg-php qui n’est plus maintenu au (...)
Sur d’autres sites (13762)
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ffmpeg GRAY16 stream over network
28 novembre 2023, par Norbert P.Im working in a school project where we need to use depth cameras. The camera produces color and depth (in other words 16bit grayscale image). We decided to use ffmpeg, as later on compression could be very useful. For now we got some basic stream running form one PC to other. These settings include :


- 

- rtmp
- flv as container
- pixel format AV_PIX_FMT_YUV420P
- codec AV_CODEC_ID_H264










The problem we are having is with grayscale image. Not every codec is able to cope with this format, so as not every protocol able to work with given codec. I got some settings "working" but receiver side is just stuck on avformat_open_input() method.
I have also tested it with commandline where ffmpeg is listening for connection and same happens.


I include a minimum "working" example of client code. Server can be tested with "ffmpeg.exe -f apng -listen 1 -i rtmp ://localhost:9999/stream/stream1 -c copy -f apng -listen 1 rtmp ://localhost:2222/live/l" or code below. I get no warnings, ffmpeg is newest version installed with "vcpkg install —triplet x64-windows ffmpeg[ffmpeg,ffprobe,zlib]" on windows or packet manager on linux.


The question : Did I miss something ? How do I get it to work ? If you have any better ideas I would very gladly consider them. In the end I need 16 bits of lossless transmission, could be split between channels etc. which I also tried with same effect.


Client code that would have camera and connect to server :


extern "C" {
#include <libavutil></libavutil>opt.h>
#include <libavcodec></libavcodec>avcodec.h>
#include <libavutil></libavutil>channel_layout.h>
#include <libavutil></libavutil>common.h>
#include <libavformat></libavformat>avformat.h>
#include <libavcodec></libavcodec>avcodec.h>
#include <libavutil></libavutil>imgutils.h>
}

int main() {

 std::string container = "apng";
 AVCodecID codec_id = AV_CODEC_ID_APNG;
 AVPixelFormat pixFormat = AV_PIX_FMT_GRAY16BE;

 AVFormatContext* format_ctx;
 AVCodec* out_codec;
 AVStream* out_stream;
 AVCodecContext* out_codec_ctx;
 AVFrame* frame;
 uint8_t* data;

 std::string server = "rtmp://localhost:9999/stream/stream1";

 int width = 1280, height = 720, fps = 30, bitrate = 1000000;

 //initialize format context for output with flv and no filename
 avformat_alloc_output_context2(&format_ctx, nullptr, container.c_str(), server.c_str());
 if (!format_ctx) {
 return 1;
 }

 //AVIOContext for accessing the resource indicated by url
 if (!(format_ctx->oformat->flags & AVFMT_NOFILE)) {
 int avopen_ret = avio_open(&format_ctx->pb, server.c_str(),
 AVIO_FLAG_WRITE);// , nullptr, nullptr);
 if (avopen_ret < 0) {
 fprintf(stderr, "failed to open stream output context, stream will not work\n");
 return 1;
 }
 }


 const AVCodec* tmp_out_codec = avcodec_find_encoder(codec_id);
 //const AVCodec* tmp_out_codec = avcodec_find_encoder_by_name("hevc");
 out_codec = const_cast(tmp_out_codec);
 if (!(out_codec)) {
 fprintf(stderr, "Could not find encoder for '%s'\n",
 avcodec_get_name(codec_id));

 return 1;
 }

 out_stream = avformat_new_stream(format_ctx, out_codec);
 if (!out_stream) {
 fprintf(stderr, "Could not allocate stream\n");
 return 1;
 }

 out_codec_ctx = avcodec_alloc_context3(out_codec);

 const AVRational timebase = { 60000, fps };
 const AVRational dst_fps = { fps, 1 };
 av_log_set_level(AV_LOG_VERBOSE);
 //codec_ctx->codec_tag = 0;
 //codec_ctx->codec_id = codec_id;
 out_codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO;
 out_codec_ctx->width = width;
 out_codec_ctx->height = height;
 out_codec_ctx->gop_size = 1;
 out_codec_ctx->time_base = timebase;
 out_codec_ctx->pix_fmt = pixFormat;
 out_codec_ctx->framerate = dst_fps;
 out_codec_ctx->time_base = av_inv_q(dst_fps);
 out_codec_ctx->bit_rate = bitrate;
 //if (fctx->oformat->flags & AVFMT_GLOBALHEADER)
 //{
 // codec_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 //}

 out_stream->time_base = out_codec_ctx->time_base; //will be set afterwards by avformat_write_header to 1/1000

 int ret = avcodec_parameters_from_context(out_stream->codecpar, out_codec_ctx);
 if (ret < 0)
 {
 fprintf(stderr, "Could not initialize stream codec parameters!\n");
 return 1;
 }

 AVDictionary* codec_options = nullptr;
 av_dict_set(&codec_options, "tune", "zerolatency", 0);

 // open video encoder
 ret = avcodec_open2(out_codec_ctx, out_codec, &codec_options);
 if (ret < 0)
 {
 fprintf(stderr, "Could not open video encoder!\n");
 return 1;
 }
 av_dict_free(&codec_options);

 out_stream->codecpar->extradata_size = out_codec_ctx->extradata_size;
 out_stream->codecpar->extradata = static_cast(av_mallocz(out_codec_ctx->extradata_size));
 memcpy(out_stream->codecpar->extradata, out_codec_ctx->extradata, out_codec_ctx->extradata_size);

 av_dump_format(format_ctx, 0, server.c_str(), 1);

 frame = av_frame_alloc();

 int sz = av_image_get_buffer_size(pixFormat, width, height, 32);
#ifdef _WIN32
 data = (uint8_t*)_aligned_malloc(sz, 32);
 if (data == NULL)
 return ENOMEM;
#else
 ret = posix_memalign(reinterpret_cast(&data), 32, sz);
#endif
 av_image_fill_arrays(frame->data, frame->linesize, data, pixFormat, width, height, 32);
 frame->format = pixFormat;
 frame->width = width;
 frame->height = height;
 frame->pts = 1;
 if (avformat_write_header(format_ctx, nullptr) < 0) //Header making problems!!!
 {
 fprintf(stderr, "Could not write header!\n");
 return 1;
 }

 printf("stream time base = %d / %d \n", out_stream->time_base.num, out_stream->time_base.den);

 double inv_stream_timebase = (double)out_stream->time_base.den / (double)out_stream->time_base.num;
 printf("Init OK\n");
 /* Init phase end*/
 int dts = 0;
 int frameNo = 0;

 while (true) {
 //Fill dummy frame with something
 for (int y = 0; y < height; y++) {
 uint16_t color = ((y + frameNo) * 256) % (256 * 256);
 for (int x = 0; x < width; x++) {
 data[x+y*width] = color;
 }
 }

 memcpy(frame->data[0], data, 1280 * 720 * sizeof(uint16_t));
 AVPacket* pkt = av_packet_alloc();

 int ret = avcodec_send_frame(out_codec_ctx, frame);
 if (ret < 0)
 {
 fprintf(stderr, "Error sending frame to codec context!\n");
 return ret;
 }
 while (ret >= 0) {
 ret = avcodec_receive_packet(out_codec_ctx, pkt);
 if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
 break;
 else if (ret < 0) {
 fprintf(stderr, "Error during encoding\n");
 break;
 }
 pkt->dts = dts;
 pkt->pts = dts;
 dts += 33;
 av_write_frame(format_ctx, pkt);
 frameNo++;
 av_packet_unref(pkt);
 }
 printf("Streamed %d frames\n", frameNo);
 }
 return 0;
}


And part of server that should receive. code where is stops and waits


extern "C" {
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libavformat></libavformat>avio.h>
}

int main() {
 AVFormatContext* fmt_ctx = NULL;
 av_log_set_level(AV_LOG_VERBOSE);
 AVDictionary* options = nullptr;
 av_dict_set(&options, "protocol_whitelist", "file,udp,rtp,tcp,rtmp,rtsp,hls", 0);
 av_dict_set(&options, "timeout", "500000", 0); // Timeout in microseconds 

//Next Line hangs 
 int ret = avformat_open_input(&fmt_ctx, "rtmp://localhost:9999/stream/stream1", NULL, &options);
 if (ret != 0) {
 fprintf(stderr, "Could not open RTMP stream\n");
 return -1;
 }

 // Find the first video stream
 ret = avformat_find_stream_info(fmt_ctx, nullptr);
 if (ret < 0) {
 return ret;
 }
 //...
} 



Edit :
I tried to just create a animated png and tried to stream that from the console to another console window to avoid any programming mistakes on my side. It was the same, I just could not get 16 PNG encoded stream to work. I hung trying to receive and closed when the file ended with in total zero frames received.


I managed to get other thing working :
To not encode gray frames with YUV420, I installed ffmpeg with libx264 support (was thinking is the same as H264, which in code is, but it adds support to new pixel formats). Used H264 again but with GRAY8 with doubled image width and reconstructing the image on the other side.


Maybe as a side note, I could not get any other formats to work. Is "flv" the only option here ? Could I get more performance if I changed it to... what ?


-
How to debug ffmpeg reliability for long running rtsp streams
13 septembre 2022, par MarkI have a long running ffmpeg background process that "watches" an rtsp stream and takes snapshots every 7 minutes.


It's being run like this


C:\Windows\System32\cmd.exe /c C:\ffmpeg\bin\ffmpeg.exe -nostdin -rtsp_transport tcp -y -timeout 5000000 -i rtsp://someurl -q:v 1 -an -vf fps=0.002381,scale="1280:720" -strftime 1 -f image2 C:\somelocalfolder\%Y-%m-%d_%H-%M-%S.jpg > c:\ffmpeglog.txt 2>&1


This process runs for days but intermittently, for hours at a time, seems to miss taking snapshots, until eventually it starts to take them again - then fail again, etc. The logs at info level are not helpful. I checked the stream during times when it was not taking snapshots and the stream was up. What's happening here ? How can I debug this ?


Below is an image of succesfull snapshots per hour. There should always be between 8 and 9.



Logs look like this


ffmpeg version 2022-03-31-git-e301a24fa1-full_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers
 built with gcc 11.2.0 (Rev7, Built by MSYS2 project)
 configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint
 libavutil 57. 24.101 / 57. 24.101
 libavcodec 59. 25.100 / 59. 25.100
 libavformat 59. 20.101 / 59. 20.101
 libavdevice 59. 6.100 / 59. 6.100
 libavfilter 8. 29.100 / 8. 29.100
 libswscale 6. 6.100 / 6. 6.100
 libswresample 4. 6.100 / 4. 6.100
 libpostproc 56. 5.100 / 56. 5.100
Input #0, rtsp, from 'rtsp://somerul':
 Metadata:
 title : HIK Media Server V4.21.005
 comment : HIK Media Server Session Description : standard
 Duration: N/A, start: 0.033000, bitrate: N/A
 Stream #0:0: Video: h264 (High), yuv420p(progressive), 704x576, 30 tbr, 90k tbn
Stream mapping:
 Stream #0:0 -> #0:0 (h264 (native) -> mjpeg (native))
[swscaler @ 000002a1c2c20680] [swscaler @ 000002a1c2c2e0c0] deprecated pixel format used, make sure you did set range correctly
[swscaler @ 000002a1c2c20680] [swscaler @ 000002a1c2c67c40] deprecated pixel format used, make sure you did set range correctly
[swscaler @ 000002a1c2c20680] [swscaler @ 000002a1c2cc6700] deprecated pixel format used, make sure you did set range correctly
Output #0, image2, to 'C:\somelocalfolder\Temp\stream_2\StreamedImages\%Y-%m-%d_%H-%M-%S.jpg':
 Metadata:
 title : HIK Media Server V4.21.005
 comment : HIK Media Server Session Description : standard
 encoder : Lavf59.20.101
 Stream #0:0: Video: mjpeg, yuvj420p(pc, progressive), 1280x720, q=2-31, 200 kb/s, 0.0024 fps, 0.0024 tbn
 Metadata:
 encoder : Lavc59.25.100 mjpeg
 Side data:
 cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: N/A
frame= 1 fps=0.0 q=0.0 size=N/A time=00:00:00.00 bitrate=N/A speed= 0x 


Update
I got some trace logs. The ffmpeg seems to fail silently at some point and stop taking snapshots.


After about 3 million log lines (which is really only a couple of hours in my case) I get the following


rtsp://192.168.15.195:554/streaming/channels/904: Unknown error


But ffmpeg silently continues. Here is a bit more of the log


[Parsed_fps_0 @ 00000248e7d50e40] Read frame with in pts 1074443040, out pts 28
[Parsed_fps_0 @ 00000248e7d50e40] Dropping frame with pts 28
frame= 28 fps=0.0 q=1.0 size=N/A time=03:08:59.77 bitrate=N/A speed=0.95x 
[rtsp @ 00000248e765cf00] tcp_read_packet:
[h264 @ 00000248e7d59880] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 3
[rtsp @ 00000248e765cf00] ret=-138 c=24 [$]
rtsp://192.168.15.195:554/streaming/channels/904: Unknown error
[Parsed_fps_0 @ 00000248e7d50e40] Read frame with in pts 1074446100, out pts 28
[Parsed_fps_0 @ 00000248e7d50e40] Dropping frame with pts 28
frame= 28 fps=0.0 q=1.0 size=N/A time=03:08:59.77 bitrate=N/A speed=0.95x 
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=696
[rtsp @ 00000248e765cf00] Sending:
GET_PARAMETER rtsp://192.168.15.195:554/streaming/channels/904 RTSP/1.0

CSeq: 402

User-Agent: Lavf59.20.101

Session: 931848797

Authorization: Digest username="******", realm="709382dda4ccb674edf093d3", nonce="13fca62fc", uri="rtsp://192.168.15.195:554/streaming/channels/904", response="74341df9611f0ac3dc247b402424735b", algorithm="MD5"



--
[NULL @ 00000248e7662640] nal_unit_type: 7(SPS), nal_ref_idc: 3
[NULL @ 00000248e7662640] nal_unit_type: 8(PPS), nal_ref_idc: 3
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=756
[Parsed_fps_0 @ 00000248e7d50e40] Read frame with in pts 1074449070, out pts 28
[Parsed_fps_0 @ 00000248e7d50e40] Dropping frame with pts 28
[Parsed_fps_0 @ 00000248e7d50e40] Read frame with in pts 1074449070, out pts 28
[Parsed_fps_0 @ 00000248e7d50e40] Dropping frame with pts 28
frame= 28 fps=0.0 q=1.0 size=N/A time=03:08:59.77 bitrate=N/A speed=0.949x 
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
frame= 28 fps=0.0 q=1.0 size=N/A time=03:08:59.77 bitrate=N/A speed=0.949x 
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1228
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
[NULL @ 00000248e7662640] reference count 1 overflow
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=804
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
[NULL @ 00000248e7662640] illegal memory management control operation 11
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=836


Basically it appears an issue of ffmpeg silently failing. If it crashed, my software could detect it and I could rerun it, but if it fails silently like this, I need another solution.


-
Streaming RTP with ffmpeg and node.js to voip phone
5 juillet 2023, par Nik HendricksI 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
100and180responses 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
200response 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
ffplayon. 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
ffmpegto stream my audio with rtp I notice that it sees the file format as thispcm_alaw, 8000 Hz, mono, s16, 64 kb/sMy 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.


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