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• AppRTC : Google’s WebRTC test app and its parameters

  23 juillet 2014, par silvia

  If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC : AppRTC.

  When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

  We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

  Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

  Here are my favourite parameters :
  

  • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
  • stereo=true : turns on stereo audio
  • debug=loopback : connect to yourself (great to check your own firewalls)
  • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

  For example, here’s how a stereo, HD loopback test would look like : https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

  This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery :

  • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
  • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
  • tp=[password] : password for the TURN server
  • audio=true&video=false : audio-only call
  • audio=false : video-only call
  • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
  • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
  • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
  • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
  • dtls=false : disable datagram transport layer security
  • dscp=true : enable DSCP
  • ipv6=true : enable IPv6

  AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

  Have fun playing with the main and always up-to-date WebRTC application : AppRTC.

  UPDATE 12 May 2014

  AppRTC now also supports the following bitrate controls :

  • arbr=[bitrate] : set audio receive bitrate
  • asbr=[bitrate] : set audio send bitrate
  • vsbr=[bitrate] : set video receive bitrate
  • vrbr=[bitrate] : set video send bitrate

  Example usage : https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true

• FFMPEG error while receiving rtsp stream

  8 avril 2014, par Pawel Rutka

  I got error with receiving rtsp stream from IPCam Edimax IC-3030 and I don't know what to do. Can anyone help me or show me a way to solution ?

  /home/prog12# ffplay "rtsp://192.168.1.7/ipcam_h264.sdp"
  
  ffplay version 2.1.4 Copyright (c) 2003-2014 the FFmpeg developers
  
  built on Mar 22 2014 18:16:53 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)
  
  configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-      libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable- libxvid --enable-x11grab --enable-libvpx --enable-libmp3lame
  
  
  
  ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Access denied
  
  
  
   [rtsp @ 0xb0c006c0] UDP timeout, retrying with TCPsq=    0B f=0/0   
  
   [rtsp @ 0xb0c006c0] method PAUSE failed: 501 Not Implemented
  
   [rtsp @ 0xb0c006c0] Could not find codec parameters for stream 0 (Video: h264): unspecified size
  
   Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
   Input #0, rtsp, from 'rtsp://192.168.1.7/ipcam_h264.sdp':
  
   Metadata:
  
  title           : IPCam
  
   Duration: N/A, bitrate: 64 kb/s
  
  Stream #0:0: Video: h264, 90k tbr, 90k tbn, 180k tbc
  
  Stream #0:1: Audio: pcm_alaw, 8000 Hz, 1 channels, s16, 64 kb/s
  
  SDL_OpenAudio (1 channels): 
  
  No more channel combinations to try, audio open failed
  
  [rtsp @ 0xb0c006c0] UDP timeout, retrying with TCPsq=    0B f=0/0   
  
  [rtsp @ 0xb0c006c0] method PAUSE failed: 501 Not Implemented
  
  [rtsp @ 0xb0c006c0] UDP timeout, retrying with TCPsq=    0B f=0/0   
  
  [rtsp @ 0xb0c006c0] method PAUSE failed: 501 Not Implementedf=0/0   
  
  [rtsp @ 0xb0c006c0] UDP timeout, retrying with TCPsq=    0B f=0/0   
  
  [rtsp @ 0xb0c006c0] method PAUSE failed: 501 Not Implemented
  
  nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
  nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
  nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
  nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
  nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   

  Thats the output.

  EDIT

  ps ax| grep -i pulse 
  
  8904 ?        S<l></l>usr/bin/pulseaudio --start --log-target=syslog
  
  10369 pts/1    S+     0:00 grep --color=auto -i pulse

  And for new command :

  sudo ffplay -video_size 640x480 "rtsp://192.168.1.7/ipcam_h264.sdp"
  
  ffplay version 2.1.4 Copyright (c) 2003-2014 the FFmpeg developers
  
  built on Mar 22 2014 18:16:53 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)
  
  configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --enable-libvpx --enable-libmp3lame
  
  
  
  Option video_size not found.aq=    0KB vq=    0KB sq=    0B f=0/0   

• running multiple FFmpeg instances to encode udp stream gives error

  24 mars 2014, par user2854116

  Please someone help me to fix the problem below

  I am trying to encode multiple udp streams in HLS format. I run one by one instance (process) of encoding on the system. The application is working fine and I was able to encode, receive and play the streams using a player running on a STB (Set Top Box). The encoding was OK till running the 5 th instance. When I started running the 6th encoding instance on the system, the stream output starts freezing in all the 6 encoded outputs.

  I am using a Intel Xeon based Server with 24 cores and 32GB of RAM.

  Here is my ffmpeg build info.

  ffmpeg
  ffmpeg version N-61057-gec6d043 Copyright (c) 2000-2014 the FFmpeg developers
  built on Mar 4 2014 05:33:48 with gcc 4.6 (Debian 4.6.3-1)
  configuration : —prefix=/root/ffmpeg-static/64bit —extra-cflags='-I/root/ffmpeg-static/64bit/include -static' —extra-ldflags='-L/root/ffmpeg-static/64bit/lib -static' —extra-libs='-lxml2 -lexpat -lfreetype' —enable-static —disable-shared —disable-ffserver —disable-doc —enable-bzlib —enable-zlib —enable-postproc —enable-runtime-cpudetect —enable-libx264 —enable-gpl —enable-libtheora —enable-libvorbis —enable-libmp3lame —enable-gray —enable-libass —enable-libfreetype —enable-libopenjpeg —enable-libspeex —enable-libvo-aacenc —enable-libvo-amrwbenc —enable-version3 —enable-libvpx

  I am using the below command to encode one channel :

  ffmpeg -loglevel info -i "udp://@239.1.1.9:1009?fifo_size=1000000&overrun_nonfatal=1" -map 0:p:1:0 -map 0:p:1:1 -vcodec h264 -acodec aac -profile main -preset fast -g 12 -r 23.98 -s 256x144 -aspect 16:9 -b:v 166k -minrate 166k -maxrate 166k -bufsize 166k -vol 512 -b:a 128k -ac 2 -ar 48000 -strict experimental -flags -global_header -f ssegment -segment_time 10 -segment_wrap 30000 -segment_list Food_256x144_main_394.m3u8 -segment_list_flags +live -segment_list_size 6 -segment_format mpegts Food_256x144_main_394-%05d.ts -map 0:p:2:0 -map 0:p:2:1 -vcodec h264 -acodec aac -profile main -preset fast -g 12 -r 23.98 -s 640x360 -aspect 16:9 -b:v 722k -minrate 722k -maxrate 722k -bufsize 722k -vol 512 -b:a 128k -ac 2 -ar 48000 -strict experimental -flags -global_header -f ssegment -segment_time 10 -segment_wrap 30000 -segment_list Food_640x360_main_950.m3u8 -segment_list_flags +live -segment_list_size 6 -segment_format mpegts Food_640x360_main_950-%05d.ts -map 0:p:2:0 -map 0:p:2:1 -vcodec h264 -acodec aac -profile main -preset fast -g 12 -r 23.98 -s 960x540 -aspect 16:9 -b:v 1472k -minrate 1472k -maxrate 1472k -bufsize 1472k -vol 512 -b:a 128k -ac 2 -ar 48000 -strict experimental -flags -global_header -f ssegment -segment_time 10 -segment_wrap 30000 -segment_list Food_960x540_main_1700.m3u8 -segment_list_flags +live -segment_list_size 6 -segment_format mpegts Food_960x540_main_1700-%05d.ts -map 0:p:2:0 -map 0:p:2:1 -vcodec h264 -acodec aac -profile main -preset fast -g 12 -r 23.98 -s 1280x720 -aspect 16:9 -b:v 2772k -minrate 2772k -maxrate 2772k -bufsize 2772k -vol 512 -b:a 128k -ac 2 -ar 48000 -strict experimental -flags -global_header -f ssegment -segment_time 10 -segment_wrap 30000 -segment_list Food_1280x720_main_3000.m3u8 -segment_list_flags +live -segment_list_size 6 -segment_format mpegts Food_1280x720_main_3000-%05d.ts

  The sample error messages I am getting after adding the 6 th encoding instance is as below ( I have pasted only few lines of error message for reference as the log message is too long )

  • [list=]
    2563489 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 1 5
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 10 6
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 33 7
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] ac-tex damaged at 37 8
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] slice mismatch
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] ac-tex damaged at 10 10
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 5 11
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 2 12
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 1 13
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 16 14
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 4 15
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 13 16
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 6 17
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 24 18
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 25 19
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 4 20
    2563490 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 28 21
    2563491 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] slice mismatch
    2563491 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] ac-tex damaged at 38 23
    2563491 [main] DEBUG com.ivz.streamer.FFmpegStreamer - Stream from process : [mpeg2video @ 0x24d2100] 00 motion_type at 14 24