| Message ID | cb9f2532-877b-4679-a47e-bdfff3184932@gmail.com |
|---|---|
| State | New |
| Headers | show |
| Series | [FFmpeg-devel,v2,1/2] avformat/rtsp: Support mode field of Transport header being sent in upper case | expand |
| Context | Check | Description |
|---|---|---|
| andriy/configure_x86 | warning | Failed to apply patch |
| yinshiyou/configure_loongarch64 | warning | Failed to apply patch |
On Mon, Jan 15, 2024 at 10:37:25PM +0200, Paul Orlyk wrote: > Fixes server compatibility issues with rtspclientsink GStreamer plugin. > > > From specification: > RFC 7826 "Real-Time Streaming Protocol Version 2.0" (https://datatracker.ietf.org/doc/html/rfc7826), section 18.54: > mode: The mode parameter indicates the methods to be supported for > this session. The currently defined valid value is "PLAY". If > not provided, the default is "PLAY". The "RECORD" value was > defined in RFC 2326; in this specification, it is unspecified > but reserved. RECORD and other values may be specified in the > future. > RFC 2326 "Real Time Streaming Protocol (RTSP)" (https://datatracker.ietf.org/doc/html/rfc2326), section 12.39: > mode: > The mode parameter indicates the methods to be supported for > this session. Valid values are PLAY and RECORD. If not > provided, the default is PLAY. > > mode=receive was always like this, from the initial commit 'a8ad6ffa rtsp: Add listen mode'. > > For comparison, Wowza was used to push RTSP stream to. Both GStreamer and FFmpeg had no issues. > Here is the capture of Wowza responding to SETUP request: > 200 OK > CSeq: 3 > Server: Wowza Streaming Engine 4.8.26+4 build20231212155517 > Cache-Control: no-cache > Expires: Mon, 15 Jan 2024 19:40:31 GMT > Transport: RTP/AVP/UDP;unicast;client_port=11640-11641;mode=record;source=172.17.0.2;server_port=6976-6977 > Date: Mon, 15 Jan 2024 19:40:31 GMT > Session: 1401457689;timeout=60 > > Test setup: > Server: ffmpeg -loglevel trace -y -rtsp_flags listen -i rtsp://0.0.0.0:30800/live.stream t.mp4 > FFmpeg client: ffmpeg -re -i "Big Buck Bunny - FULL HD 30FPS.mp4" -c:v libx264 -f rtsp rtsp://127.0.0.1:30800/live.stream > GStreamer client: gst-launch-1.0 videotestsrc is-live=true pattern=smpte ! queue ! videorate ! videoscale ! video/x-raw,width=640,height=360,framerate=60/1 ! timeoverlay font-desc="Sans, 84" halignment=center valignment=center ! queue ! videoconvert ! tee name=t t. ! x264enc bitrate=9000 pass=cbr speed-preset=ultrafast byte-stream=false key-int-max=15 threads=1 ! video/x-h264,profile=baseline ! queue ! rsink. audiotestsrc ! voaacenc ! queue ! rsink. t. ! queue ! autovideosink rtspclientsink name=rsink location=rtsp://localhost:30800/live.stream > > Test results: > modified FFmpeg client -> stock server : ok > stock FFmpeg client -> modified server : ok > modified FFmpeg client -> modified server : ok > GStreamer client -> modified server : ok > > Signed-off-by: Paul Orlyk <paul.orlyk@gmail.com> > --- > libavformat/rtspdec.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) Thats a nice commit message will apply thx [...]
diff --git a/libavformat/rtspdec.c b/libavformat/rtspdec.c index 39fd92fb66..d6a223cbc6 100644 --- a/libavformat/rtspdec.c +++ b/libavformat/rtspdec.c @@ -303,7 +303,7 @@ static int rtsp_read_setup(AVFormatContext *s, char* host, char *controlurl) rtsp_st->interleaved_min = request.transports[0].interleaved_min; rtsp_st->interleaved_max = request.transports[0].interleaved_max; snprintf(responseheaders, sizeof(responseheaders), "Transport: " - "RTP/AVP/TCP;unicast;mode=receive;interleaved=%d-%d" + "RTP/AVP/TCP;unicast;mode=record;interleaved=%d-%d" "\r\n", request.transports[0].interleaved_min, request.transports[0].interleaved_max); } else { @@ -333,7 +333,7 @@ static int rtsp_read_setup(AVFormatContext *s, char* host, char *controlurl) localport = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle); snprintf(responseheaders, sizeof(responseheaders), "Transport: " - "RTP/AVP/UDP;unicast;mode=receive;source=%s;" + "RTP/AVP/UDP;unicast;mode=record;source=%s;" "client_port=%d-%d;server_port=%d-%d\r\n", host, request.transports[0].client_port_min, request.transports[0].client_port_max, localport,