我使用 node.js 从 WebRTC 服务器（我使用 mediasoup）获取 RTP 流，并从流中获取解密的 RTP 数据包原始数据。我想将此 RTP 数据转发到 ffmpeg。我创建了描述音频和视频流的 SDP 文件，并通过 UDP 发送数据包。社会民主党：
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 33400 RTP/AVP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=mid:audio
a=recvonly
m=video 33402 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=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=mid:video
a=recvonly
a=rtcp-mux
命令：ffmpeg -loglevel debug -analyzeduration 2147483647 -probesize 2147483647 -protocol_whitelist file,crypto,udp,rtp -re -vcodec vp8 -acodec opus -i test.sdp -vcodec h264 -acodec aac -y output.mp4
日志：
ffmpeg version 3.2
Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-11)
configuration: --prefix=/opt/kaltura/ffmpeg-3.2 --libdir=/opt/kaltura/ffmpeg-3.2/lib --shlibdir=/opt/kaltura/ffmpeg-3.2/lib --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC -I/opt/kaltura/include' --extra-ldflags=-L/opt/kaltura/lib --disable-devices --enable-bzlib --enable-libgsm --enable-libmp3lame --enable-libschroedinger --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libx265 --enable-avisynth --enable-libxvid --enable-filter=movie --enable-avfilter --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libvpx --enable-libspeex --enable-libass --enable-postproc --enable-pthreads --enable-static --enable-shared --enable-gpl --disable-debug --disable-optimizations --enable-gpl --enable-pthreads --enable-swscale --enable-vdpau --enable-bzlib --disable-devices --enable-filter=movie --enable-version3 --enable-indev=lavfi --enable-x11grab
libavutil 55. 34.100 / 55. 34.100
libavcodec 57. 64.100 / 57. 64.100
libavformat 57. 56.100 / 57. 56.100
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
Splitting the commandline.
Reading option '-loglevel' ...
matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-analyzeduration' ...
matched as AVOption 'analyzeduration' with argument '2147483647'.
Reading option '-probesize' ...
matched as AVOption 'probesize' with argument '2147483647'.
Reading option '-protocol_whitelist' ...
matched as AVOption 'protocol_whitelist' with argument 'file,crypto,udp,rtp'.
Reading option '-re' ...
matched as option 're' (read input at native frame rate) with argument '1'.
Reading option '-vcodec' ...
matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'vp8'.
Reading option '-acodec' ...
matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'opus'.
Reading option '-i' ... matched as input file with argument 'test.sdp'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'h264'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option 'output.mp4' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file test.sdp.
Applying option re (read input at native frame rate) with argument 1.
Applying option vcodec (force video codec ('copy' to copy stream)) with argument vp8.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument opus.
Successfully parsed a group of options.
Opening an input file: test.sdp.
[sdp @ 0xb1ef00] Format sdp probed with size=2048 and score=50
[sdp @ 0xb1ef00] audio codec set to: (null)
[sdp @ 0xb1ef00] audio samplerate set to: 44100
[sdp @ 0xb1ef00] audio channels set to: 1
[sdp @ 0xb1ef00] video codec set to: (null)
[udp @ 0xb21940] end receive buffer size reported is 131072
[udp @ 0xb21660] end receive buffer size reported is 131072
[sdp @ 0xb1ef00] setting jitter buffer size to 500
[udp @ 0xb21da0] end receive buffer size reported is 131072
[udp @ 0xb22060] end receive buffer size reported is 131072
[sdp @ 0xb1ef00] setting jitter buffer size to 500
[sdp @ 0xb1ef00] Before avformat_find_stream_info() pos: 889 bytes read:889 seeks:0 nb_streams:2
[vp8 @ 0xb27600] Header size larger than data provided
Last message repeated 2 times
[sdp @ 0xb1ef00] Non-increasing DTS in stream 1: packet 2 with DTS 0, packet 3 with DTS 0
[vp8 @ 0xb27600] Header size larger than data provided
... repeats many times until I kill the socket ...
Last message repeated 1 times
[sdp @ 0xb1ef00] Non-increasing DTS in stream 1: packet 273 with DTS 553050, packet 274 with DTS 553050
[vp8 @ 0xb27600] Header size larger than data provided
received id=7199daf55e496b370e36cd1d25b1ef5b9dff6858 type=bye
PeerConnection close. id=7199daf55e496b370e36cd1d25b1ef5b9dff6858
-- PeerConnection.closed, err: undefined
-- peers in the room = 0
[sdp @ 0xb1ef00] decoding for stream 1 failed
[sdp @ 0xb1ef00] Could not find codec parameters for stream 1 (Video: vp8, 1 reference frame, yuv420p): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[sdp @ 0xb1ef00] After avformat_find_stream_info() pos: 889 bytes read:889 seeks:0 frames:584
Input #0, sdp, from 'test.sdp':
Metadata:
title : 7199daf55e496b370e36cd1d25b1ef5b9dff6858
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0, 309, 1/90000: Audio: opus, 48000 Hz, mono, fltp
Stream #0:1, 275, 1/90000: Video: vp8, 1 reference frame, yuv420p, 90k tbr, 90k tbn, 90k tbc
Successfully opened the file.
Parsing a group of options: output file output.mp4.
Applying option vcodec (force video codec ('copy' to copy stream)) with argument h264.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument aac.
Successfully parsed a group of options.
Opening an output file: output.mp4.
Matched encoder 'libx264' for codec 'h264'.
[file @ 0xbc56e0]
Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 1 logical cores
[graph 0 input from stream 0:1 @ 0xb1eca0]
Setting 'video_size' to value '0x0'
[buffer @ 0xbc54e0]
Unable to parse option value "0x0" as image size
[graph 0 input from stream 0:1 @ 0xb1eca0]
Setting 'pix_fmt' to value '0'
[graph 0 input from stream 0:1 @ 0xb1eca0]
Setting 'time_base' to value '1/90000'
[graph 0 input from stream 0:1 @ 0xb1eca0] Setting 'pixel_aspect' to value '0/1'
[graph 0 input from stream 0:1 @ 0xb1eca0] Setting 'sws_param' to value 'flags=2'
[graph 0 input from stream 0:1 @ 0xb1eca0] Setting 'frame_rate' to value '90000/1'
[buffer @ 0xbc54e0] Unable to parse option value "0x0" as image size
[buffer @ 0xbc54e0] Error setting option video_size to value 0x0.
[graph 0 input from stream 0:1 @ 0xb1eca0] Error applying options to the filter.
Error opening filters!
[AVIOContext @ 0xbc57c0] Statistics: 0 seeks, 0 writeouts
[AVIOContext @ 0xb1f8c0]
Statistics: 889 bytes read, 0 seeks
如您所见，在日志的开头，SDP 在没有识别编解码器的情况下进行了解析：
Opening an input file: test.sdp.
[sdp @ 0xb1ef00] Format sdp probed with size=2048 and score=50
[sdp @ 0xb1ef00] audio codec set to: (null)
[sdp @ 0xb1ef00] audio samplerate set to: 44100
[sdp @ 0xb1ef00] audio channels set to: 1
[sdp @ 0xb1ef00] video codec set to: (null)
然后它试图从套接字读取数据包。只有当我关闭套接字时，ffmpeg 才会继续解析 SDP，这一次找到了正确的编解码器：
Opening an input file: test.sdp.
[sdp @ 0xb1ef00] Format sdp probed with size=2048 and score=50
[sdp @ 0xb1ef00] audio codec set to: (null)
[sdp @ 0xb1ef00] audio samplerate set to: 44100
[sdp @ 0xb1ef00] audio channels set to: 1
[sdp @ 0xb1ef00] video codec set to: (null)
我怀疑“不增加 DTS”和“标头大小大于提供的数据”错误是由于使用了错误的编解码器而导致的数据包解析错误引起的。
我检查了 SDP 顺序，它似乎与我的其他示例相同。
有人可以提出解释吗？
顺便说一句，音频只能正常工作，但我想这是因为 OPUS 的简单性。
谢谢。