I get RTP stream from WebRTC server (I used mediasoup) using node.js and I get the decrypted RTP packets raw data from the stream. I want to forward this RTP data to ffmpeg and from there I can save it to file, or push it as RTMP stream to other media servers. I guess that the best way would be to create SDP file that describes both the audio and video streams and send the packets through new sockets.
The ffmpeg command is:
ffmpeg -loglevel debug -protocol_whitelist file,crypto,udp,rtp -re -vcodec libvpx -acodec opus -i test.sdp -vcodec libx264 -acodec aac -y output.mp4
I tried to send the packets through UDP:
v=0
o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
c=IN IP4 192.168.193.182
t=0 0
m=audio 33301 RTP/AVP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=sendrecv
m=video 33302 RTP/AVP 100
a=rtpmap:100 /VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=sendrecv
But I always get (removed the boring parts):
Opening an input file: test.sdp.
[sdp @ 0x103dea0]
Format sdp probed with size=2048 and score=50
[sdp @ 0x103dea0] audio codec set to: (null)
[sdp @ 0x103dea0] audio samplerate set to: 44100
[sdp @ 0x103dea0] audio channels set to: 1
[sdp @ 0x103dea0] video codec set to: (null)
[udp @ 0x10402e0] end receive buffer size reported is 131072
[udp @ 0x10400c0] end receive buffer size reported is 131072
[sdp @ 0x103dea0] setting jitter buffer size to 500
[udp @ 0x1040740] bind failed: Address already in use
[AVIOContext @ 0x1046980] Statistics: 473 bytes read, 0 seeks
test.sdp: Invalid data found when processing input
Note that I get it even if I don't open socket at all or send anything to this port, as if the ffmpeg itself tries to open these ports more than once.
I tried also to open two (video and audio) TCP servers and define SDP with TCP:
v=0
o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
c=IN IP4 192.168.193.182
t=0 0
m=audio 33301 TCP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=setup:active
a=connection:new
a=sendrecv
m=video 33302 TCP 100
a=rtpmap:100 /VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=setup:active
a=connection:new
a=sendrecv
However I don't see any incoming connection into my TCP servers and I get the following from ffmpeg:
Opening an input file: test.sdp.
[sdp @ 0xdddea0]
Format sdp probed with size=2048 and score=50
[sdp @ 0xdddea0]
audio codec set to: (null)
[sdp @ 0xdddea0]
audio samplerate set to: 44100
[sdp @ 0xdddea0] audio channels set to: 1
[sdp @ 0xdddea0] video codec set to: (null)
[udp @ 0xde02e0] end receive buffer size reported is 131072
[udp @ 0xde00c0] end receive buffer size reported is 131072
[sdp @ 0xdddea0] setting jitter buffer size to 500
[udp @ 0xde0740] end receive buffer size reported is 131072
[udp @ 0xde0180] end receive buffer size reported is 131072
[sdp @ 0xdddea0] setting jitter buffer size to 500
[sdp @ 0xdddea0] Before avformat_find_stream_info() pos: 593 bytes read:593 seeks:0 nb_streams:2
[libvpx @ 0xdeea80] v1.3.0
[libvpx @ 0xdeea80] --target=x86_64-linux-gcc --enable-pic --disable-install-srcs --as=nasm --enable-shared --prefix=/usr --libdir=/usr/lib64
[sdp @ 0xdddea0] Could not find codec parameters for stream 1 (Video: vp8, 1 reference frame, none): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[sdp @ 0xdddea0] After avformat_find_stream_info() pos: 593 bytes read:593 seeks:0 frames:0
Input #0, sdp, from 'test.sdp':
Metadata:
title : 7199daf55e496b370e36cd1d25b1ef5b9dff6858
Duration: N/A, bitrate: N/A
Stream #0:0, 0, 1/90000: Audio: opus, 48000 Hz, mono, fltp
Stream #0:1, 0, 1/90000: Video: vp8, 1 reference frame, none, 90k tbr, 90k tbn, 90k tbc
Successfully opened the file.
Parsing a group of options: output file output.mp4.
Successfully parsed a group of options.
Opening an output file: output.mp4.
[file @ 0xde3660] Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 1 logical cores
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'sample_fmt' to value 'fltp'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 0:0 @ 0xde3940] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x4
[audio format for output stream 0:0 @ 0xe37900] Setting 'sample_fmts' to value 'fltp'
[audio format for output stream 0:0 @ 0xe37900] Setting 'sample_rates' to value '96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
[AVFilterGraph @ 0xde0220] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
Output #0, mp4, to 'output.mp4':
Metadata:
title :
7199daf55e496b370e36cd1d25b1ef5b9dff6858
encoder :
Lavf57.56.100
Stream #0:0
, 0, 1/48000
: Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, mono, fltp, delay 1024, 69 kb/s
Metadata:
encoder :
Lavc57.64.100 aac
Stream mapping:
Stream #0:0 -> #0:0 (opus (native) -> aac (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
test.sdp: Connection timed out
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[output stream 0:0 @ 0xde3b40] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
[aac @ 0xde2b00] Trying to remove 1024 samples, but the queue is empty
[aac @ 0xde2b00] Trying to remove 1024 more samples than there are in the queue
[mp4 @ 0xe6a540] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
[mp4 @ 0xe6a540] Encoder did not produce proper pts, making some up.
[aac @ 0xde2b00] Trying to remove 1024 samples, but the queue is empty
[aac @ 0xde2b00] Trying to remove 1024 more samples than there are in the queue
size= 1kB time=00:00:00.04 bitrate= 157.9kbits/s speed=0.00426x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3268.000000%
Input file #0 (test.sdp):
Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples);
Input stream #0:1 (video): 0 packets read (0 bytes);
Total: 0 packets (0 bytes) demuxed
Output file #0 (output.mp4):
Output stream #0:0 (audio): 0 frames encoded (0 samples); 2 packets muxed (25 bytes);
Total: 2 packets (25 bytes) muxed
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0xde37a0] Statistics: 30 seeks, 25 writeouts
[aac @ 0xde2b00] Qavg: 47249.418
[AVIOContext @ 0xde6980] Statistics: 593 bytes read, 0 seeks
Note to the "Connection timed out" in the log above.
I guess that both my SDPs are wrong, any suggestions?
Alternatives to SDP are also most welcomed.
c=IN IP4 192.168.193.182
Is that your local IP from which your own Node UDP/TCP server is listening for the connection from ffmpeg?
m=audio 33301 RTP/AVP 111
Why 33301? I hope that is not the same port as the one used by mediasoup to communicate with the remote browser (if so, obviously you¡ll get the "address already in use" error)...
a=rtpmap:111 /opus/48000
That's wrong format. Remove the first "/".
Remove all the a=rtcp-fb lines. I don't think ffmpeg supports any of them at all.
Same for video.
I've solved my problem, my setup :
kms(rtpEndpoint)-->.sdp--->ffmpeg--->nginx-rtmp--->vlc
Problem was in sdp. sdp should be in proper format, as sdp answer. Use identical sdp content in file(ffmpeg sdp input file) as well as in media server(kms in my case). IP should be nginx server ip.(works in my case).
来源：https://stackoverflow.com/questions/42900514/stream-rtp-to-ffmpeg-using-sdp