Hello
I am struggling to read a multicast audio RTP stream controlled by RTSP. As
soon as I launch the ffplay or ffmpeg command, the RTP traffic starts
(tcpdump) on port 5004/UDP (as advertised in the SDP file), so it should
read the stream correctly but I finally get a time-out instead (and empty
out file). The very same RTSP URL can be read without any issue with VLC or
with rtpdump+sox.
$ ffmpeg -y -rtsp_transport udp_multicast -i "rtsp://
192.168.2.148:554/by-name/AES67-stream (on hasseb-AoE-F8-82)" -vn -f s24le
-ar 48000 -c:a pcm_s24be out.raw
ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --disable-debug
--enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb
--enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb
--enable-libpulse --enable-libfreetype --enable-gnutl
s --enable-libdav1d --enable-libx264 --enable-libx265 --enable-libfdk-aac
--enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx
--enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr
--enable-libxvid --enable-libvidstab --enable-libtheora --en
able-libwavpack --enable-libopenjpeg --enable-libgsm --enable-nvenc
--enable-libzimg --enable-libaom
libavutil 56. 31.100 / 56. 31.100
libavcodec 58. 54.100 / 58. 54.100
libavformat 58. 29.100 / 58. 29.100
libavdevice 58. 8.100 / 58. 8.100
libavfilter 7. 57.100 / 7. 57.100
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, rtsp, from 'rtsp://192.168.2.148:554/by-name/AES67-stream (on
hasseb-AoE-F8-82)':
Metadata:
title : AES67-stream (on hasseb-AoE-F8-82) streamed by
"hasseb"
Duration: N/A, bitrate: 2304 kb/s
Stream #0:0: Audio: pcm_s24be, 48000 Hz, stereo, s32 (24 bit), 2304 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s24be (native) -> pcm_s24be (native))
Press [q] to stop, [?] for help
rtsp://192.168.2.148:554/by-name/AES67-stream (on hasseb-AoE-F8-82):
Connection timed out
Output #0, s24le, to 'out.raw':
Metadata:
title : AES67-stream (on hasseb-AoE-F8-82) streamed by
"hasseb"
encoder : Lavf58.29.100
Stream #0:0: Audio: pcm_s24be, 48000 Hz, stereo, s32 (24 bit), 2304 kb/s
Metadata:
encoder : Lavc58.54.100 pcm_s24be
size= 0kB time=00:00:00.00 bitrate=N/A speed= 0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames
parameters if used)
The SDP file is :
v=0
o=- 254522267104 0 IN IP4 192.168.2.148
s=AES67-stream (on hasseb-AoE-F8-82) streamed by "hasseb"
t=0 0
a=clock-domain:PTPv2 0
a=recvonly
m=audio 5004 RTP/AVP 98
c=IN IP4 239.123.123.123/255
a=rtpmap:98 L24/48000/2
a=sync-time:0
a=framecount:48
a=source-filter: incl IN IP4 239.123.123.123 192.168.2.148
a=ts-refclk:ptp=IEEE1588-2008:00-10-4b-ff-fe-2e-f8-82:domain-nmbr=0
a=mediaclk:direct=0
a=ptime:1
Comparing those files, we see that ffmpeg detects the correct
audio settings (pcm_s24be, 48000 Hz, stereo, s32 (24 bit)) so I cannot
figure out why it can't read the data stream? I have tried to increase the
timeout but that did not help.
Any help would be appreciated. Thank you.
Yannick
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