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  • Gestion générale des documents

    13 mai 2011, par

    MédiaSPIP ne modifie jamais le document original mis en ligne.
    Pour chaque document mis en ligne il effectue deux opérations successives : la création d’une version supplémentaire qui peut être facilement consultée en ligne tout en laissant l’original téléchargeable dans le cas où le document original ne peut être lu dans un navigateur Internet ; la récupération des métadonnées du document original pour illustrer textuellement le fichier ;
    Les tableaux ci-dessous expliquent ce que peut faire MédiaSPIP (...)

  • MediaSPIP v0.2

    21 juin 2013, par

    MediaSPIP 0.2 est la première version de MediaSPIP stable.
    Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Comme pour la version précédente, 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 (...)

  • Mise à disposition des fichiers

    14 avril 2011, par

    Par défaut, lors de son initialisation, MediaSPIP ne permet pas aux visiteurs de télécharger les fichiers qu’ils soient originaux ou le résultat de leur transformation ou encodage. Il permet uniquement de les visualiser.
    Cependant, il est possible et facile d’autoriser les visiteurs à avoir accès à ces documents et ce sous différentes formes.
    Tout cela se passe dans la page de configuration du squelette. Il vous faut aller dans l’espace d’administration du canal, et choisir dans la navigation (...)

Sur d’autres sites (5513)

  • nginx RTMP to HLS : FFMPG error when trying multiple bitrate output [on hold]

    28 mai 2014, par user3685074

    I’m currently trying to convert my RTMP Livestream into a HLS with 3 quality-settings.

    I followed this guide

    I’ve compiled my own FFMPEG and it’s working if I just convert 1 file.
    It seems libx264 isn’t able to do multiple encodings at the same time ?

    I’m using these command :

           exec /usr/local/bin/ffmpeg -i rtmp://localhost/src/$name
           -c:a libfdk_aac -b:a 32k   -c:v libx264 -b:v 128K -f flv rtmp://localhost/hls/$name_low
           -c:a libfdk_aac -b:a 64k   -c:v libx264 -b:v 256K -f flv rtmp://localhost/hls/$name_mid
           -c:a libfdk_aac -b:a 128k  -c:v libx264 -b:v 512K -f flv rtmp://localhost/hls/$name_hi  2>>/tmp/ffmpeg.log;

    this is the output :

       ffmpeg version N-63519-g61917a1 Copyright (c) 2000-2014 the FFmpeg developers
         built on May 28 2014 18:06:42 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
         configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab --enable-libvpx --enable-libmp3lame --enable-librtmp --enable-libspeex --enable-libfdk_aac
         libavutil      52. 87.100 / 52. 87.100
         libavcodec     55. 65.100 / 55. 65.100
         libavformat    55. 41.100 / 55. 41.100
         libavdevice    55. 13.101 / 55. 13.101
         libavfilter     4.  5.100 /  4.  5.100
         libswscale      2.  6.100 /  2.  6.100
         libswresample   0. 19.100 /  0. 19.100
         libpostproc    52.  3.100 / 52.  3.100
       Metadata:
         Server                NGINX RTMP (github.com/arut/nginx-rtmp-module)
         width                 1280.00
         height                720.00
         displayWidth          1280.00
         displayHeight         720.00
         duration              0.00
         framerate             25.00
         fps                   25.00
         videodatarate         390.00
         videocodecid          0.00
         audiodatarate         27.00
         audiocodecid          11.00
       Input #0, flv, from 'rtmp://localhost/src/test':
         Metadata:
           Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
           displayWidth    : 1280
           displayHeight   : 720
           fps             : 25
           profile         :
           level           :
         Duration: 00:00:00.00, start: 0.080000, bitrate: N/A
           Stream #0:0: Video: h264 (High), yuv420p, 1280x720, 399 kb/s, 25 fps, 25 tbr, 1k tbn, 50 tbc
           Stream #0:1: Audio: speex, 16000 Hz, mono, s16, 27 kb/s
       [libx264 @ 0x5260380] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
       [libx264 @ 0x5260380] profile High, level 3.1
       [libx264 @ 0x5260380] 264 - core 142 r2431 f23da7c - 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 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=24 lookahead_threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=128 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
       [libx264 @ 0x525a920] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
       Output #0, flv, to 'rtmp://localhost/hls/test_low':
         Metadata:
           Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
           displayWidth    : 1280
           displayHeight   : 720
           fps             : 25
           profile         :
           level           :
           Stream #0:0: Video: h264 (libx264), yuv420p, 1280x720, q=-1--1, 128 kb/s, 25 fps, 90k tbn, 25 tbc
           Metadata:
             encoder         : Lavc55.65.100 libx264
           Stream #0:1: Audio: aac (libfdk_aac), 16000 Hz, mono, s16, 32 kb/s
           Metadata:
             encoder         : Lavc55.65.100 libfdk_aac
       Output #1, flv, to 'rtmp://localhost/hls/test_mid':
         Metadata:
           Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
           displayWidth    : 1280
           displayHeight   : 720
           fps             : 25
           profile         :
           level           :
           Stream #1:0: Video: h264, yuv420p, 1280x720, q=-1--1, 256 kb/s, 25 fps, 90k tbn, 25 tbc
           Metadata:
             encoder         : Lavc55.65.100 libx264
           Stream #1:1: Audio: aac, 16000 Hz, mono, s16
           Metadata:
             encoder         : Lavc55.65.100 libfdk_aac
       Output #2, flv, to 'rtmp://localhost/hls/test_hi':
         Metadata:
           Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
           displayWidth    : 1280
           displayHeight   : 720
           fps             : 25
           profile         :
           level           :
           Stream #2:0: Video: h264, yuv420p, 1280x720, q=-1--1, 25 fps, 90k tbn, 25 tbc
           Metadata:
             encoder         : Lavc55.65.100 libx264
           Stream #2:1: Audio: aac, 16000 Hz, mono, s16
           Metadata:
             encoder         : Lavc55.65.100 libfdk_aac
       Stream mapping:
         Stream #0:0 -> #0:0 (h264 -> libx264)
         Stream #0:1 -> #0:1 (libspeex -> libfdk_aac)
         Stream #0:0 -> #1:0 (h264 -> libx264)
         Stream #0:1 -> #1:1 (libspeex -> libfdk_aac)
         Stream #0:0 -> #2:0 (h264 -> libx264)
         Stream #0:1 -> #2:1 (libspeex -> libfdk_aac)
       Error while opening encoder for output stream #1:0 - maybe incorrect parameters such as bit_rate, rate, width or height

    I hope you can help me and sorry for my bad english.

    Greetz
    Kevin

  • nginx RTMP to HLS : FFMPG error when trying multiple bitrate output [closed]

    28 mai 2014, par user3685074

    I’m currently trying to convert my RTMP Livestream into a HLS with 3 quality-settings.

    I followed this guide

    I’ve compiled my own FFMPEG and it’s working if I just convert 1 file.
    It seems libx264 isn’t able to do multiple encodings at the same time ?

    I’m using these command :

           exec /usr/local/bin/ffmpeg -i rtmp://localhost/src/$name
           -c:a libfdk_aac -b:a 32k   -c:v libx264 -b:v 128K -f flv rtmp://localhost/hls/$name_low
           -c:a libfdk_aac -b:a 64k   -c:v libx264 -b:v 256K -f flv rtmp://localhost/hls/$name_mid
           -c:a libfdk_aac -b:a 128k  -c:v libx264 -b:v 512K -f flv rtmp://localhost/hls/$name_hi  2>>/tmp/ffmpeg.log;

    this is the output :

       ffmpeg version N-63519-g61917a1 Copyright (c) 2000-2014 the FFmpeg developers
         built on May 28 2014 18:06:42 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
         configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab --enable-libvpx --enable-libmp3lame --enable-librtmp --enable-libspeex --enable-libfdk_aac
         libavutil      52. 87.100 / 52. 87.100
         libavcodec     55. 65.100 / 55. 65.100
         libavformat    55. 41.100 / 55. 41.100
         libavdevice    55. 13.101 / 55. 13.101
         libavfilter     4.  5.100 /  4.  5.100
         libswscale      2.  6.100 /  2.  6.100
         libswresample   0. 19.100 /  0. 19.100
         libpostproc    52.  3.100 / 52.  3.100
       Metadata:
         Server                NGINX RTMP (github.com/arut/nginx-rtmp-module)
         width                 1280.00
         height                720.00
         displayWidth          1280.00
         displayHeight         720.00
         duration              0.00
         framerate             25.00
         fps                   25.00
         videodatarate         390.00
         videocodecid          0.00
         audiodatarate         27.00
         audiocodecid          11.00
       Input #0, flv, from 'rtmp://localhost/src/test':
         Metadata:
           Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
           displayWidth    : 1280
           displayHeight   : 720
           fps             : 25
           profile         :
           level           :
         Duration: 00:00:00.00, start: 0.080000, bitrate: N/A
           Stream #0:0: Video: h264 (High), yuv420p, 1280x720, 399 kb/s, 25 fps, 25 tbr, 1k tbn, 50 tbc
           Stream #0:1: Audio: speex, 16000 Hz, mono, s16, 27 kb/s
       [libx264 @ 0x5260380] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
       [libx264 @ 0x5260380] profile High, level 3.1
       [libx264 @ 0x5260380] 264 - core 142 r2431 f23da7c - 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 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=24 lookahead_threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=128 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
       [libx264 @ 0x525a920] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
       Output #0, flv, to 'rtmp://localhost/hls/test_low':
         Metadata:
           Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
           displayWidth    : 1280
           displayHeight   : 720
           fps             : 25
           profile         :
           level           :
           Stream #0:0: Video: h264 (libx264), yuv420p, 1280x720, q=-1--1, 128 kb/s, 25 fps, 90k tbn, 25 tbc
           Metadata:
             encoder         : Lavc55.65.100 libx264
           Stream #0:1: Audio: aac (libfdk_aac), 16000 Hz, mono, s16, 32 kb/s
           Metadata:
             encoder         : Lavc55.65.100 libfdk_aac
       Output #1, flv, to 'rtmp://localhost/hls/test_mid':
         Metadata:
           Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
           displayWidth    : 1280
           displayHeight   : 720
           fps             : 25
           profile         :
           level           :
           Stream #1:0: Video: h264, yuv420p, 1280x720, q=-1--1, 256 kb/s, 25 fps, 90k tbn, 25 tbc
           Metadata:
             encoder         : Lavc55.65.100 libx264
           Stream #1:1: Audio: aac, 16000 Hz, mono, s16
           Metadata:
             encoder         : Lavc55.65.100 libfdk_aac
       Output #2, flv, to 'rtmp://localhost/hls/test_hi':
         Metadata:
           Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
           displayWidth    : 1280
           displayHeight   : 720
           fps             : 25
           profile         :
           level           :
           Stream #2:0: Video: h264, yuv420p, 1280x720, q=-1--1, 25 fps, 90k tbn, 25 tbc
           Metadata:
             encoder         : Lavc55.65.100 libx264
           Stream #2:1: Audio: aac, 16000 Hz, mono, s16
           Metadata:
             encoder         : Lavc55.65.100 libfdk_aac
       Stream mapping:
         Stream #0:0 -> #0:0 (h264 -> libx264)
         Stream #0:1 -> #0:1 (libspeex -> libfdk_aac)
         Stream #0:0 -> #1:0 (h264 -> libx264)
         Stream #0:1 -> #1:1 (libspeex -> libfdk_aac)
         Stream #0:0 -> #2:0 (h264 -> libx264)
         Stream #0:1 -> #2:1 (libspeex -> libfdk_aac)
       Error while opening encoder for output stream #1:0 - maybe incorrect parameters such as bit_rate, rate, width or height

    I hope you can help me and sorry for my bad english.

    Greetz
    Kevin

  • FFplay requesting video via RTSP :// but receiving on multicast address

    28 mai 2014, par DavidG

    First of all, I apologize for how long the supporting information will be in this post. This is my first post on this forum.

    My issue is I need to run the command line version of ffmpeg to capture a video stream. However, as a proof of concept I’m first attempting to capture and view the video using ffplay (BTW, I have not had any success using ffmpeg or ffprobe). I’m running the ffplay command to read video from a Coretec video encoder which has multicast enabled.

    Unicast address:   172.30.18.50
    Multicast address: 239.130.18.50:4002

    My question is how can I request the Unicast address, but receive the video on the multicast address ? (BTW, the ffplay operation does not work even if I replace the Unicast address with the Multicast address below)

    NOTE : After looking at the Wireshark trace, I see the video data has GSMTAP in the protocol column. When I do "ffmpeg -protocols : I see there is a Decoder "gsm" which decodes raw gsm. however, when I use ffplay -f gsm ... I get "Protocol not found".

    I am able to use VLC to view the video using the following command :

    VLC rtsp://172.30.18.50

    It appears from the Wireshark trace that the session is initiated on the Unicast address, but the video is streamed on the Multicast address. VLC is able to determine this and perform the appropriate operation. I don’t know what to add to ffplay to let it know that another stream will be carrying the video.

    I am UNABLE to perform the following ffplay commands (none of them work) :

    ffplay -v debug rtsp://172.30.18.50
    ffplay -v debug -rtsp_transport udp rtsp://172.30.18.50
    ffplay -v debug -rtsp_transport udp_multicast rtsp://172.30.18.50

    NOTE : I am able to get ffplay to launch, but the video is garbled badly. Maybe this bit of information will ring a bell for someone ? The command I used was :

    ffplay -v debug -i udp://239.130.18.50:4002?sources=172.30.18.50

    The version of ffplay I’m using is :

    ffplay version N-63439-g96470ca Copyright (c) 2003-2014 the FFmpeg developers
     built on May 25 2014 22:09:07 with gcc 4.8.2 (GCC)
     configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
    isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
    le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
    e --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-
    libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libope
    njpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsox
    r --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 --enable-libxvid --enable-decklink --enable-zlib
     libavutil      52. 86.100 / 52. 86.100
     libavcodec     55. 65.100 / 55. 65.100
     libavformat    55. 41.100 / 55. 41.100
     libavdevice    55. 13.101 / 55. 13.101
     libavfilter     4.  5.100 /  4.  5.100
     libswscale      2.  6.100 /  2.  6.100
     libswresample   0. 19.100 /  0. 19.100
     libpostproc    52.  3.100 / 52.  3.100

    The debug output for ffplay -v debug rtsp ://172.30.18.50 is :

    [rtsp @ 0000000002a8be80] SDP:=    0KB vq=    0KB sq=    0B f=0/0
    v=0
    o=- 1 1 IN IP4 50.18.30.172
    s=Test
    a=type:broadcast
    t=0 0
    c=IN IP4 239.130.18.50/63
    m=video 4002 RTP/AVP 96
    a=rtpmap:96 MP4V-ES/90000
    a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8
    A058BA9860FA616087828307a=control:track1

    [rtsp @ 0000000002a8be80] video codec set to: mpeg4
    [udp @ 0000000002a8bac0] end receive buffer size reported is 65536
    [udp @ 0000000002aa1600] end receive buffer size reported is 65536
    [rtsp @ 0000000002a8be80] Nonmatching transport in server reply/0
    rtsp://172.30.18.50: Invalid data found when processing input

    And the Wireshark trace output is :

    OPTIONS rtsp://172.30.18.50:554 RTSP/1.0
    CSeq: 1
    User-Agent: Lavf55.41.100

    RTSP/1.0 200 OK
    CSeq: 1
    Public: DESCRIBE, SETUP, TEARDOWN, PLAY

    DESCRIBE rtsp://172.30.18.50:554 RTSP/1.0
    Accept: application/sdp
    CSeq: 2
    User-Agent: Lavf55.41.100

    RTSP/1.0 200 OK
    CSeq: 2 Content-Type: application/sdp
    Content-Length: 270

    v=0
    o=- 1 1 IN IP4 50.18.30.172
    s=Test
    a=type:broadcast
    t=0 0
    c=IN IP4 239.130.18.50/63
    m=video 4002 RTP/AVP 96
    a=rtpmap:96 MP4V-ES/90000
    a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8A058BA9860FA616087828307a=control:track1

    SETUP rtsp://172.30.18.50:554 RTSP/1.0
    Transport: RTP/AVP/UDP;unicast;client_port=9574-9575
    CSeq: 3
    User-Agent: Lavf55.41.100

    RTSP/1.0 200 OK
    CSeq: 3
    Session: test
    Transport: RTP/AVP;multicast;destination=;port=4002-4003;ttl=63

    The debug output for ffplay -v debug -rtsp_transport udp rtsp ://172.30.18.50 is :

    [rtsp @ 0000000002c5c0a0] SDP:=    0KB vq=    0KB sq=    0B f=0/0
    v=0
    o=- 1 1 IN IP4 50.18.30.172
    s=Test
    a=type:broadcast
    t=0 0
    c=IN IP4 239.130.18.50/63
    m=video 4002 RTP/AVP 96
    a=rtpmap:96 MP4V-ES/90000
    a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8
    A058BA9860FA616087828307a=control:track1


    [rtsp @ 0000000002c5c0a0] video codec set to: mpeg4
    [udp @ 0000000002c62420] end receive buffer size reported is 65536
    [udp @ 0000000002c726a0] end receive buffer size reported is 65536
    [rtsp @ 0000000002c5c0a0] Nonmatching transport in server reply/0
    rtsp://172.30.18.50: Invalid data found when processing input

    And the Wireshark trace output is :

    OPTIONS rtsp://172.30.18.50:554 RTSP/1.0
    CSeq: 1
    User-Agent: Lavf55.41.100

    RTSP/1.0 200 OK
    CSeq: 1
    Public: DESCRIBE, SETUP, TEARDOWN, PLAY

    DESCRIBE rtsp://172.30.18.50:554 RTSP/1.0
    Accept: application/sdp
    CSeq: 2
    User-Agent: Lavf55.41.100

    RTSP/1.0 200 OK
    CSeq: 2
    Content-Type: application/sdp
    Content-Length: 270

    v=0
    o=- 1 1 IN IP4 50.18.30.172
    s=Test
    a=type:broadcast
    t=0 0
    c=IN IP4 239.130.18.50/63
    m=video 4002 RTP/AVP 96
    a=rtpmap:96 MP4V-ES/90000 a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8A058BA9860FA616087828307a=control:track1

    SETUP rtsp://172.30.18.50:554 RTSP/1.0
    Transport: RTP/AVP/UDP;unicast;client_port=22332-22333
    CSeq: 3
    User-Agent: Lavf55.41.100

    RTSP/1.0 200 OK
    CSeq: 3
    Session: test
    Transport: RTP/AVP;multicast;destination=239.130.18.50;port=4002-4003;ttl=63

    The debug output for ffplay -v debug -rtsp_transport udp_multicast is :

    [rtsp @ 00000000002fc100] SDP:=    0KB vq=    0KB sq=    0B f=0/0
    v=0
    o=- 1 1 IN IP4 50.18.30.172
    s=Test
    a=type:broadcast
    t=0 0
    c=IN IP4 239.130.18.50/63
    m=video 4002 RTP/AVP 96
    a=rtpmap:96 MP4V-ES/90000
    a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8
    A058BA9860FA616087828307a=control:track1

    [rtsp @ 00000000002fc100] video codec set to: mpeg4
       nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0

    And the Wireshark trace output is :

    OPTIONS rtsp://172.30.18.50:554
    RTSP/1.0
    CSeq: 1
    User-Agent: Lavf55.41.100

    RTSP/1.0 200 OK
    CSeq: 1
    Public: DESCRIBE, SETUP, TEARDOWN, PLAY

    DESCRIBE rtsp://172.30.18.50:554 RTSP/1.0
    Accept: application/sdp
    CSeq: 2
    User-Agent: Lavf55.41.100

    RTSP/1.0 200 OK
    CSeq: 2
    Content-Type: application/sdp
    Content-Length: 270

    v=0
    o=- 1 1 IN IP4 50.18.30.172
    s=Test
    a=type:broadcast
    t=0 0
    c=IN IP4 239.130.18.50/63
    m=video 4002 RTP/AVP 96
    a=rtpmap:96 MP4V-ES/90000
    a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C8F8A058BA9860FA616087828307a=control:track1

    SETUP rtsp://172.30.18.50:554 RTSP/1.0
    Transport: RTP/AVP/UDP;multicast
    CSeq: 3
    User-Agent: Lavf55.41.100

    Thank you in advance to whomever is willing to tackle this.
    - DavidG