hi,
in short our rtsp server not working:-( but neither: - gst-rtsp-server, - vlc's rtsp server, - mplayer's live555. in a bit more detail, the setup: - we've got high resolution 2MP, 3MP, 5MP and high fps 12-25 fps ip cameras which has buildin rtsp server and can stream the video in rtsp. - our network is not perfect. there are a few pc which are behind 2-3-4 switch (ie. the shortest path from one server to the client goes through 2-3-4 switch). what is always working: - if we connect from a pc directly to the ip cameras('s rtsp server) then we can get the proper video stream even with gstreamer, vlc etc. what is not always working: - if we create an rtsp server on pc (eg Wim's gst-rtsp-server) which do nothing just re-transmit the rtp stream from the camera with this pipeline on the server side: rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! rtph264depay ! rtph264pay name=pay0 pt=96 and on the client side with this: gst-launch rtspsrc location="rtsp://192.168.209.200:8554/test" ! decodebin2 ! xvimagesink then on some network segment we've got gray images most of the time. we try to replace both the client and then server to vlc and mplayer (just to know whether it's a gstreamer bug or not) and neither combination working. but if we connect directly to the camera it's always working with all clients. an even strange thing if we start to stream on the server side from an mkv file (with the high resolution high fps file) than the same thing happened on the client side. so the problem is not the source of the rtps server. one small tipp: if we lower the mtu to 300 (!) then it's working, but obviously we can't use this settings in a real environment. so what can be the problem? any help would be useful. -- Levente "Si vis pacem para bellum!" _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
On 07/22/2011 03:43 PM, Farkas Levente wrote:
> hi, > in short our rtsp server not working:-( > but neither: > - gst-rtsp-server, > - vlc's rtsp server, > - mplayer's live555. > > in a bit more detail, the setup: > - we've got high resolution 2MP, 3MP, 5MP and high fps 12-25 fps ip > cameras which has buildin rtsp server and can stream the video in rtsp. > - our network is not perfect. there are a few pc which are behind 2-3-4 > switch (ie. the shortest path from one server to the client goes through > 2-3-4 switch). > > what is always working: > - if we connect from a pc directly to the ip cameras('s rtsp server) > then we can get the proper video stream even with gstreamer, vlc etc. > > what is not always working: > - if we create an rtsp server on pc (eg Wim's gst-rtsp-server) which do > nothing just re-transmit the rtp stream from the camera with this > pipeline on the server side: > > rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! > rtph264depay ! rtph264pay name=pay0 pt=96 > > and on the client side with this: > gst-launch rtspsrc location="rtsp://192.168.209.200:8554/test" ! > decodebin2 ! xvimagesink > > then on some network segment we've got gray images most of the time. > we try to replace both the client and then server to vlc and mplayer > (just to know whether it's a gstreamer bug or not) and neither > combination working. but if we connect directly to the camera it's > always working with all clients. > > an even strange thing if we start to stream on the server side from an > mkv file (with the high resolution high fps file) than the same thing > happened on the client side. so the problem is not the source of the > rtps server. > > one small tipp: if we lower the mtu to 300 (!) then it's working, but > obviously we can't use this settings in a real environment. > > so what can be the problem? any help would be useful. more info. if we also test with h263pay. is we leave the ffenc_h263p encoders bitrate at the default value (300000) than one udp packet size is about ~300bytes and in this case the video is perfect. if we rise the bitrate to 1800000 then one udp packet size is not enough for one gstbuffer. in this case the video is unusable on the receiver side. so is there any rtp (de)fragmentation algorithm in gstreamer? how does it works? we wireshark the rtp stream from the ip camera and it contains fragmented rtp packet. while we wireshark the rtp stream from gst's rtsp server the it's just a bunch of udp packets and wireshirk do not recognize them as fragmented rtp packets. is there any docs about how rtp fragmentation should have to do? what is implemented in gstreamer? fyi the encoder pipeline: rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! decodebin2 ! queue2 ! ffenc_h263p bitrate=1800000 ! rtph263ppay name=pay0 pt=96 thanks in advance. regards. -- Levente "Si vis pacem para bellum!" _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
>> what is not always working:
>> - if we create an rtsp server on pc (eg Wim's gst-rtsp-server) which do >> nothing just re-transmit the rtp stream from the camera with this >> pipeline on the server side: >> >> rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! >> rtph264depay ! rtph264pay name=pay0 pt=96 You need to re-insert the configuration data in the stream. See the config-interval parameter in the rtph264pay element. Some encoders pass sps/pps out of band (not in the rtp stream) and only in the RTSP negociation. If you want to send the data back out; re-insert the config data in the stream. Note that if you start the decoder before the encoder; you might have missed this effect because (IIRC), the payloader will send sps/pps once at startup. > more info. if we also test with h263pay. is we leave the ffenc_h263p > encoders bitrate at the default value (300000) than one udp packet size > is about ~300bytes and in this case the video is perfect. if we rise the > bitrate to 1800000 then one udp packet size is not enough for one > gstbuffer. in this case the video is unusable on the receiver side. > > so is there any rtp (de)fragmentation algorithm in gstreamer? how does > it works? we wireshark the rtp stream from the ip camera and it contains > fragmented rtp packet. while we wireshark the rtp stream from gst's rtsp > server the it's just a bunch of udp packets and wireshirk do not > recognize them as fragmented rtp packets. > is there any docs about how rtp fragmentation should have to do? > what is implemented in gstreamer? RTP packets the data under the MTU. It is reassembed on the receiving side. Can you verify that you do not have any packet loss, that would also explain the 'gray' images. > > fyi the encoder pipeline: > rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! > decodebin2 ! queue2 ! ffenc_h263p bitrate=1800000 ! rtph263ppay > name=pay0 pt=96 IIRC, only mpeg4videopay and rtph264pay (mpeg4 and h264) use the config data. However, H263 has different payload schemes (1998/2000) and two different payloaders (h263pay and h263ppay); make certain those are matches for your tests. -- br, marc The best approach for you requires two steps. The first step is reading the Manual. Wolfgang Denk - u-boot mailing list _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
On 07/25/2011 02:39 PM, Marc Leeman wrote:
>>> what is not always working: >>> - if we create an rtsp server on pc (eg Wim's gst-rtsp-server) which do >>> nothing just re-transmit the rtp stream from the camera with this >>> pipeline on the server side: >>> >>> rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! >>> rtph264depay ! rtph264pay name=pay0 pt=96 > > You need to re-insert the configuration data in the stream. See the > config-interval parameter in the rtph264pay element. Some encoders > pass sps/pps out of band (not in the rtp stream) and only in the RTSP > negociation. If you want to send the data back out; re-insert the > config data in the stream. > > Note that if you start the decoder before the encoder; you might have > missed this effect because (IIRC), the payloader will send sps/pps > once at startup. this is not working. anyway we test with stream from a h264/mkv file with wim's rtsp-server but the result was the same if the video is high resolution, high fps then the client always got gray video. and in this case it's not a resend. >> more info. if we also test with h263pay. is we leave the ffenc_h263p >> encoders bitrate at the default value (300000) than one udp packet size >> is about ~300bytes and in this case the video is perfect. if we rise the >> bitrate to 1800000 then one udp packet size is not enough for one >> gstbuffer. in this case the video is unusable on the receiver side. >> >> so is there any rtp (de)fragmentation algorithm in gstreamer? how does >> it works? we wireshark the rtp stream from the ip camera and it contains >> fragmented rtp packet. while we wireshark the rtp stream from gst's rtsp >> server the it's just a bunch of udp packets and wireshirk do not >> recognize them as fragmented rtp packets. >> is there any docs about how rtp fragmentation should have to do? >> what is implemented in gstreamer? > > RTP packets the data under the MTU. It is reassembed on the receiving > side. Can you verify that you do not have any packet loss, that would > also explain the 'gray' images. we measure with iperf and there is no pocket loss on the lan:-( >> >> fyi the encoder pipeline: >> rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! >> decodebin2 ! queue2 ! ffenc_h263p bitrate=1800000 ! rtph263ppay >> name=pay0 pt=96 > > IIRC, only mpeg4videopay and rtph264pay (mpeg4 and h264) use the > config data. However, H263 has different payload schemes (1998/2000) > and two different payloaders (h263pay and h263ppay); make certain > those are matches for your tests. we need h264, and test with it, but after we've no more idea we also test with h263, but none of them working:-( it seems gstreamer do not implement rtp fragmentation which cause all of these problems. but the other problem we do not find any kind of docs on the net about how rtp fragmentation should have to right... -- Levente "Si vis pacem para bellum!" _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
In reply to this post by Farkas Levente
Il 25/07/2011 14:27, Farkas Levente ha scritto:
> On 07/22/2011 03:43 PM, Farkas Levente wrote: >> hi, >> in short our rtsp server not working:-( >> but neither: >> - gst-rtsp-server, >> - vlc's rtsp server, >> - mplayer's live555. >> >> in a bit more detail, the setup: >> - we've got high resolution 2MP, 3MP, 5MP and high fps 12-25 fps ip >> cameras which has buildin rtsp server and can stream the video in rtsp. >> - our network is not perfect. there are a few pc which are behind 2-3-4 >> switch (ie. the shortest path from one server to the client goes through >> 2-3-4 switch). >> >> what is always working: >> - if we connect from a pc directly to the ip cameras('s rtsp server) >> then we can get the proper video stream even with gstreamer, vlc etc. >> >> what is not always working: >> - if we create an rtsp server on pc (eg Wim's gst-rtsp-server) which do >> nothing just re-transmit the rtp stream from the camera with this >> pipeline on the server side: >> >> rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! >> rtph264depay ! rtph264pay name=pay0 pt=96 >> >> and on the client side with this: >> gst-launch rtspsrc location="rtsp://192.168.209.200:8554/test" ! >> decodebin2 ! xvimagesink >> >> then on some network segment we've got gray images most of the time. >> we try to replace both the client and then server to vlc and mplayer >> (just to know whether it's a gstreamer bug or not) and neither >> combination working. but if we connect directly to the camera it's >> always working with all clients. >> >> an even strange thing if we start to stream on the server side from an >> mkv file (with the high resolution high fps file) than the same thing >> happened on the client side. so the problem is not the source of the >> rtps server. >> >> one small tipp: if we lower the mtu to 300 (!) then it's working, but >> obviously we can't use this settings in a real environment. >> >> so what can be the problem? any help would be useful. > more info. if we also test with h263pay. is we leave the ffenc_h263p > encoders bitrate at the default value (300000) than one udp packet size > is about ~300bytes and in this case the video is perfect. if we rise the > bitrate to 1800000 then one udp packet size is not enough for one > gstbuffer. in this case the video is unusable on the receiver side. > > so is there any rtp (de)fragmentation algorithm in gstreamer? how does > it works? we wireshark the rtp stream from the ip camera and it contains > fragmented rtp packet. while we wireshark the rtp stream from gst's rtsp > server the it's just a bunch of udp packets and wireshirk do not > recognize them as fragmented rtp packets. > is there any docs about how rtp fragmentation should have to do? > what is implemented in gstreamer? > > fyi the encoder pipeline: > rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! > decodebin2 ! queue2 ! ffenc_h263p bitrate=1800000 ! rtph263ppay > name=pay0 pt=96 try this pipeline: rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! rtph264depay ! rtph264pay name=pay0 pt=96 let me know, Nicola > > thanks in advance. > regards. > _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
In reply to this post by Farkas Levente
Hi,
On Fri, Jul 22, 2011 at 6:43 AM, Farkas Levente <[hidden email]> wrote: > hi, > in short our rtsp server not working:-( > but neither: > - gst-rtsp-server, > - vlc's rtsp server, > - mplayer's live555. > > in a bit more detail, the setup: > - we've got high resolution 2MP, 3MP, 5MP and high fps 12-25 fps ip > cameras which has buildin rtsp server and can stream the video in rtsp. > - our network is not perfect. there are a few pc which are behind 2-3-4 > switch (ie. the shortest path from one server to the client goes through > 2-3-4 switch). > > what is always working: > - if we connect from a pc directly to the ip cameras('s rtsp server) > then we can get the proper video stream even with gstreamer, vlc etc. > > what is not always working: > - if we create an rtsp server on pc (eg Wim's gst-rtsp-server) which do > nothing just re-transmit the rtp stream from the camera with this > pipeline on the server side: > > rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! > rtph264depay ! rtph264pay name=pay0 pt=96 > > and on the client side with this: > gst-launch rtspsrc location="rtsp://192.168.209.200:8554/test" ! > decodebin2 ! xvimagesink > > then on some network segment we've got gray images most of the time. Do you mean 100% gray images (same color for all the pixels) or a grayscale version of the images generated from the camera? > we try to replace both the client and then server to vlc and mplayer > (just to know whether it's a gstreamer bug or not) and neither > combination working. but if we connect directly to the camera it's > always working with all clients. > > an even strange thing if we start to stream on the server side from an > mkv file (with the high resolution high fps file) than the same thing > happened on the client side. so the problem is not the source of the > rtps server. > > one small tipp: if we lower the mtu to 300 (!) then it's working, but > obviously we can't use this settings in a real environment. How are you proxying the stream? Maybe you can share some code. Regards > > so what can be the problem? any help would be useful. > > > -- > Levente "Si vis pacem para bellum!" > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel > gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
In reply to this post by Mailing List SVR
Hi,
On 07/25/2011 08:10 PM, Mailing List SVR wrote: > Il 25/07/2011 14:27, Farkas Levente ha scritto: >> On 07/22/2011 03:43 PM, Farkas Levente wrote: >>> hi, >>> in short our rtsp server not working:-( >>> but neither: >>> - gst-rtsp-server, >>> - vlc's rtsp server, >>> - mplayer's live555. >>> >>> in a bit more detail, the setup: >>> - we've got high resolution 2MP, 3MP, 5MP and high fps 12-25 fps ip >>> cameras which has buildin rtsp server and can stream the video in rtsp. >>> - our network is not perfect. there are a few pc which are behind 2-3-4 >>> switch (ie. the shortest path from one server to the client goes >>> through >>> 2-3-4 switch). >>> >>> what is always working: >>> - if we connect from a pc directly to the ip cameras('s rtsp server) >>> then we can get the proper video stream even with gstreamer, vlc etc. >>> >>> what is not always working: >>> - if we create an rtsp server on pc (eg Wim's gst-rtsp-server) which do >>> nothing just re-transmit the rtp stream from the camera with this >>> pipeline on the server side: >>> >>> rtspsrc >>> location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! >>> rtph264depay ! rtph264pay name=pay0 pt=96 >>> >>> and on the client side with this: >>> gst-launch rtspsrc location="rtsp://192.168.209.200:8554/test" ! >>> decodebin2 ! xvimagesink >>> >>> then on some network segment we've got gray images most of the time. >>> we try to replace both the client and then server to vlc and mplayer >>> (just to know whether it's a gstreamer bug or not) and neither >>> combination working. but if we connect directly to the camera it's >>> always working with all clients. >>> >>> an even strange thing if we start to stream on the server side from an >>> mkv file (with the high resolution high fps file) than the same thing >>> happened on the client side. so the problem is not the source of the >>> rtps server. >>> >>> one small tipp: if we lower the mtu to 300 (!) then it's working, but >>> obviously we can't use this settings in a real environment. >>> >>> so what can be the problem? any help would be useful. >> more info. if we also test with h263pay. is we leave the ffenc_h263p >> encoders bitrate at the default value (300000) than one udp packet size >> is about ~300bytes and in this case the video is perfect. if we rise the >> bitrate to 1800000 then one udp packet size is not enough for one >> gstbuffer. in this case the video is unusable on the receiver side. >> >> so is there any rtp (de)fragmentation algorithm in gstreamer? how does >> it works? we wireshark the rtp stream from the ip camera and it contains >> fragmented rtp packet. while we wireshark the rtp stream from gst's rtsp >> server the it's just a bunch of udp packets and wireshirk do not >> recognize them as fragmented rtp packets. >> is there any docs about how rtp fragmentation should have to do? >> what is implemented in gstreamer? >> >> fyi the encoder pipeline: >> rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! >> decodebin2 ! queue2 ! ffenc_h263p bitrate=1800000 ! rtph263ppay >> name=pay0 pt=96 > > try this pipeline: > > rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! > > rtph264depay ! rtph264pay name=pay0 pt=96 > > let me know, > > Nicola > We tried, but the result is the same. In few minutes i'll make a lot of more detailed log with wireshark, samplecode and screenshoot maybe it helps to find out, what is the problem. Thanks for the help in advance. Regards. > >> >> thanks in advance. >> regards. >> > > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
In reply to this post by Marco Ballesio
Hi,
here is some detailed information about the problem: On 07/25/2011 10:22 PM, Marco Ballesio wrote: > Hi, > > On Fri, Jul 22, 2011 at 6:43 AM, Farkas Levente<[hidden email]> wrote: >> hi, >> in short our rtsp server not working:-( >> but neither: >> - gst-rtsp-server, >> - vlc's rtsp server, >> - mplayer's live555. >> >> in a bit more detail, the setup: >> - we've got high resolution 2MP, 3MP, 5MP and high fps 12-25 fps ip >> cameras which has buildin rtsp server and can stream the video in rtsp. >> - our network is not perfect. there are a few pc which are behind 2-3-4 >> switch (ie. the shortest path from one server to the client goes through >> 2-3-4 switch). >> >> what is always working: >> - if we connect from a pc directly to the ip cameras('s rtsp server) >> then we can get the proper video stream even with gstreamer, vlc etc. >> >> what is not always working: >> - if we create an rtsp server on pc (eg Wim's gst-rtsp-server) which do >> nothing just re-transmit the rtp stream from the camera with this >> pipeline on the server side: >> >> rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! >> rtph264depay ! rtph264pay name=pay0 pt=96 >> >> and on the client side with this: >> gst-launch rtspsrc location="rtsp://192.168.209.200:8554/test" ! >> decodebin2 ! xvimagesink >> >> then on some network segment we've got gray images most of the time. > Do you mean 100% gray images (same color for all the pixels) or a > grayscale version of the images generated from the camera? > >> we try to replace both the client and then server to vlc and mplayer >> (just to know whether it's a gstreamer bug or not) and neither >> combination working. but if we connect directly to the camera it's >> always working with all clients. >> >> an even strange thing if we start to stream on the server side from an >> mkv file (with the high resolution high fps file) than the same thing >> happened on the client side. so the problem is not the source of the >> rtps server. >> >> one small tipp: if we lower the mtu to 300 (!) then it's working, but >> obviously we can't use this settings in a real environment. > How are you proxying the stream? Maybe you can share some code. > client_camera.txt includes the wireshark log when we connect directly to the camera. pipeline on the client: gst-launch playbin2 uri=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main the picture is clear and good client_rtsp.txt includes the wireshark log when we connect to our rtsp server. We are using Wim's gst-rtsp-server. Here are the code from the rtsp server: #include <gst/gst.h> #include <gst/rtsp-server/rtsp-server.h> int main (int argc, char *argv[]) { GMainLoop *loop; GstRTSPServer *server; GstRTSPMediaMapping *mapping; GstRTSPMediaFactory *factory; gst_init (&argc, &argv); loop = g_main_loop_new (NULL, FALSE); server = gst_rtsp_server_new (); mapping = gst_rtsp_server_get_media_mapping (server); factory = gst_rtsp_media_factory_new (); gst_rtsp_media_factory_set_launch (factory, "( rtspsrc location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! rtph264depay ! rtph264pay name=pay0 pt=96 )"); gst_rtsp_media_factory_set_shared (factory, TRUE); gst_rtsp_media_mapping_add_factory (mapping, "/test", factory); g_object_unref (mapping); gst_rtsp_server_attach (server, NULL); g_main_loop_run (loop); return 0; } pipeline on the client: gst-launch playbin2 uri=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main the picture is gray as you see in rtsp.png if you need any more information, don't be shy to ask thanks for your help in advance best regards Peter Bocz > Regards > >> so what can be the problem? any help would be useful. >> >> >> -- >> Levente "Si vis pacem para bellum!" >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >> > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel client_camera.txt (34K) Download Attachment client_rtsp.txt (21K) Download Attachment rtsp.png (396K) Download Attachment |
Hi,
On 07/26/2011 08:51 AM, Peter BOCZ wrote: > Hi, > > here is some detailed information about the problem: > > On 07/25/2011 10:22 PM, Marco Ballesio wrote: >> Hi, >> >> On Fri, Jul 22, 2011 at 6:43 AM, Farkas Levente<[hidden email]> >> wrote: >>> hi, >>> in short our rtsp server not working:-( >>> but neither: >>> - gst-rtsp-server, >>> - vlc's rtsp server, >>> - mplayer's live555. >>> >>> in a bit more detail, the setup: >>> - we've got high resolution 2MP, 3MP, 5MP and high fps 12-25 fps ip >>> cameras which has buildin rtsp server and can stream the video in rtsp. >>> - our network is not perfect. there are a few pc which are behind 2-3-4 >>> switch (ie. the shortest path from one server to the client goes >>> through >>> 2-3-4 switch). >>> >>> what is always working: >>> - if we connect from a pc directly to the ip cameras('s rtsp server) >>> then we can get the proper video stream even with gstreamer, vlc etc. >>> >>> what is not always working: >>> - if we create an rtsp server on pc (eg Wim's gst-rtsp-server) which do >>> nothing just re-transmit the rtp stream from the camera with this >>> pipeline on the server side: >>> >>> rtspsrc >>> location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! >>> rtph264depay ! rtph264pay name=pay0 pt=96 >>> >>> and on the client side with this: >>> gst-launch rtspsrc location="rtsp://192.168.209.200:8554/test" ! >>> decodebin2 ! xvimagesink >>> >>> then on some network segment we've got gray images most of the time. >> Do you mean 100% gray images (same color for all the pixels) or a >> grayscale version of the images generated from the camera? >> > > Images look like iframes are missing, i attached one. > >>> we try to replace both the client and then server to vlc and mplayer >>> (just to know whether it's a gstreamer bug or not) and neither >>> combination working. but if we connect directly to the camera it's >>> always working with all clients. >>> >>> an even strange thing if we start to stream on the server side from an >>> mkv file (with the high resolution high fps file) than the same thing >>> happened on the client side. so the problem is not the source of the >>> rtps server. >>> >>> one small tipp: if we lower the mtu to 300 (!) then it's working, but >>> obviously we can't use this settings in a real environment. >> How are you proxying the stream? Maybe you can share some code. >> > > I attached some wireshark log maybe it is helpful. > > client_camera.txt includes the wireshark log when we connect directly > to the camera. > pipeline on the client: gst-launch playbin2 > uri=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main > the picture is clear and good > > client_rtsp.txt includes the wireshark log when we connect to our rtsp > server. We are using Wim's gst-rtsp-server. Here are the code from > the rtsp server: > > #include <gst/gst.h> > #include <gst/rtsp-server/rtsp-server.h> > > int main (int argc, char *argv[]) > { > GMainLoop *loop; > GstRTSPServer *server; > GstRTSPMediaMapping *mapping; > GstRTSPMediaFactory *factory; > > gst_init (&argc, &argv); > > loop = g_main_loop_new (NULL, FALSE); > > server = gst_rtsp_server_new (); > mapping = gst_rtsp_server_get_media_mapping (server); > factory = gst_rtsp_media_factory_new (); > gst_rtsp_media_factory_set_launch (factory, "( rtspsrc > location=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main ! > rtph264depay ! rtph264pay name=pay0 pt=96 )"); > gst_rtsp_media_factory_set_shared (factory, TRUE); > gst_rtsp_media_mapping_add_factory (mapping, "/test", factory); > g_object_unref (mapping); > gst_rtsp_server_attach (server, NULL); > > g_main_loop_run (loop); > > return 0; > } > > pipeline on the client: gst-launch playbin2 > uri=rtsp://admin:12345@192.168.209.209:554/h.264/ch1/main sorry the right pipeline is: gst-launch playbin2 uri=rtsp://192.168.209.200:8554/test > the picture is gray as you see in rtsp.png > > if you need any more information, don't be shy to ask > thanks for your help in advance > best regards > > Peter Bocz >> Regards >> >>> so what can be the problem? any help would be useful. >>> >>> >>> -- >>> Levente "Si vis pacem para bellum!" >>> _______________________________________________ >>> gstreamer-devel mailing list >>> [hidden email] >>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >>> >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel thanks for your help in advance best regards Peter Bocz _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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