If Transport: RTP/AVP/TCP

[Alex81]

Hello.

I have a RTSP-class.

In terminal, I use the command: vlc rtsp://127.0.0.1:5554/stream1.sdp

After connecting to my RTSP-server, I can get *ClientAddr.sin_addr* and
*ClientAddr.sin_port*.

Next step:

I accept SETUP from VLC-player:

"SETUP rtsp://127.0.0.1:5554/stream1.sdp/track1 RTSP/1.0\r\ nCSeq:
4\r\nUser-Agent: LibVLC/3.0.11 (LIVE555 Streaming Media
v2016.11.28)\r\n*Transport: RTP/AVP;unicast;client_port=55358*-55359\ r\
n\r\ n "

In gst-launch I set client_port = 55358:

"videotestsrc ! videoconvert ! videoscale !
video/x-raw,format=I420,width=480,height=320 ! x264enc ! video/x-h264,
stream-format=byte-stream, alignment=au, profile=main, bitrate=8000,
framerate=15/1 !  queue ! rtph264pay ! multiudpsink
clients=127.0.0.1:*55358*"

- it works fine - the video ‘videotestsrc’ is shown in the VLC player.


But, in terminal, I use the command: *vlc --rtsp-tcp*
rtsp://127.0.0.1:5554/stream1.sdp

then, I accept SETUP from the VLC player:

"SETUP rtsp://127.0.0.1:5554/stream1.sdp/track1 RTSP/1.0\r\nCSeq:
4\r\nUser-Agent: LibVLC/3.0.11 (LIVE555 Streaming Media
v2016.11.28)\r\n*Transport: RTP/AVP/TCP;unicast;interleaved=0-1*\r\n\r\n"

How, can I create gst-launch with *Transport: RTP/AVP/TCP and
interleaved=0-1*?

*Thank you for your responses.*




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