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#3 The Safest Place
16 octobre 2011, par
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ED-ME-5 1-DVD
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Revolution of Open-source and film making towards open film making
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Autres articles (94)
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Dans un premier temps il utilise le plugin "Gestion de mutualisation" -
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Sur d’autres sites (15343)
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Serving rtmp on port 1935
22 mai 2020, par JJ The SecondI've been trying to get ffmpeg to stream in rtmp but connection to port 1935 is always refused. I really don't know what else I can do to allow this connection.



Here is what specs I'm running.



- 

- Ubuntu 18.04 (tried with 19.04) however same issue - here is why I think I've made a mistake
- No Nginx installation at the moment
- FFMPEG "ffmpeg version 3.4.6-0ubuntu0.18.04.1 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)"









This is the script I run :



ffmpeg -i "test.mp4" -c:v copy -c:a copy -f flv "rtmp://127.0.0.1/stream/test"



Error I get is :



[tcp @ 0x55ff05ab8ce0] Connection to tcp://127.0.0.1:1935 failed: Connection refused



I've done some research and been across many posts about ffserver.conf and I have made those changes but still no luck. Here is my config file. I also have ran ffserver once using this config.






HTTPPort 8090
HTTPBindAddress 127.0.0.1
MaxHTTPConnections 2000
MaxClients 1000
MaxBandwidth 1000
CustomLog -

<feed>

File /tmp/feed1.ffm
FileMaxSize 200K

# Only allow connections from localhost to the feed.
ACL allow 127.0.0.1
ACL allow localhost 
ACL allow 192.168.0.0 192.168.255.255
</feed>

<stream>

# coming from live feed 'feed1'
Feed feed1.ffm

Format mpeg
AudioBitRate 32

# Number of audio channels: 1 = mono, 2 = stereo
AudioChannels 2
AudioSampleRate 44100

# Bitrate for the video stream
VideoBitRate 64

# Ratecontrol buffer size
VideoBufferSize 40

# Number of frames per second
VideoFrameRate 3


</stream>


<stream>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</stream>

# Special streams

# Server status

<stream>
Format status

ACL allow localhost
ACL allow 127.0.0.1
ACL allow 192.168.0.0 192.168.255.255

#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</stream>

<redirect>
URL http://www.ffmpeg.org/
</redirect>



Here is my ufw status :



-- ------ ----
22/tcp ALLOW Anywhere
22 ALLOW Anywhere
1935/tcp ALLOW Anywhere
22/tcp (v6) ALLOW Anywhere (v6)
22 (v6) ALLOW Anywhere (v6)
1935/tcp (v6) ALLOW Anywhere (v6)



but still nothing, I've also opened ports in iptables but no luck. Here is how this is done :



iptables -A INPUT -m state --state NEW -m tcp -p tcp --dport 1935 -j ACCEPT

and

iptables -A OUTPUT -m state --state NEW -m tcp -p tcp --dport 1935 -j ACCEPT



and still nothing, every time I run ffmpeg I get connection refused. I have previously installed nginx just to test but no luck.



What am I doing wrong here ? Isn't this port suppose to be open now ?



Thanks


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avformat/mpegtsenc : Fix mpegts_write_pes() for private_stream_2 and other types
25 avril 2021, par zheng qianavformat/mpegtsenc : Fix mpegts_write_pes() for private_stream_2 and other types
According to the PES packet definition defined in Table 2-17 of ISO_IEC_13818-1
specification, some fields like PTS/DTS or pes_extension could only appears if
the stream_id meets the condition :if (stream_id != 0xBC && // program_stream_map
stream_id != 0xBE && // padding_stream
stream_id != 0xBF && // private_stream_2
stream_id != 0xF0 && // ECM
stream_id != 0xF1 && // EMM
stream_id != 0xFF && // program_stream_directory
stream_id != 0xF2 && // DSMCC_stream
stream_id != 0xF8) // ITU-T Rec. H.222.1 type E streamAnd the following stream_id types don't have fields like PTS/DTS :
else if ( stream_id == program_stream_map
|| stream_id == private_stream_2
|| stream_id == ECM
|| stream_id == EMM
|| stream_id == program_stream_directory
|| stream_id == DSMCC_stream
|| stream_id == ITU-T Rec. H.222.1 type E stream )
for (i = 0 ; i < PES_packet_length ; i++)
PES_packet_data_byte
Current implementation skipped the check of stream_id causing some kind of
streams like private_stream_2 to be incorrectly written with actually a
private_stream_1-like PES header with PTS/DTS field. For example, Japan DTV
transmits news and alerts through ARIB superimpose that utilizes
private_stream_2 still could not be remuxed correctly for now.This patch set fixes the remuxing for private_stream_2 and
other stream_id types.Signed-off-by : zheng qian <xqq@xqq.im>
Signed-off-by : Marton Balint <cus@passwd.hu> -
Transcode HLS Segments individually using FFMPEG
27 mai 2013, par rayhI am recording a continuous, live stream to a high-bitrate HLS stream. I then want to asynchronously transcode this to different formats/bitrates. I have this working, mostly, except audio artefacts are appearing between each segment (gaps and pops).
Here is an example ffmpeg command line :
ffmpeg -threads 1 -nostdin -loglevel verbose \
-nostdin -y -i input.ts -c:a libfdk_aac \
-ac 2 -b:a 64k -y -metadata -vn output.tsInspecting an example sound file shows that there is a gap at the end of the audio :
And the start of the file looks suspiciously attenuated (although this may not be an issue) :
My suspicion is that these artefacts are happening because transcoding are occurring without the context of the stream as a whole.
Any ideas on how to convince FFMPEG to produce audio that will fit back into a HLS stream ?
** UPDATE 1 **
Here are the start/end of the original segment. As you can see, the start still appears the same, but the end is cleanly ended at 30s. I expect some degree of padding with lossy encoding, but I there is some way that HLS manages to do gapless playback (is this related to iTunes method with custom metadata ?)
** UPDATED 2 **
So, I converted both the original (128k aac in MPEG2 TS) and the transcoded (64k aac in aac/adts container) to WAV and put the two side-by-side. This is the result :
I'm not sure if this is representative of how a client will play it back, but it seems a bit odd that decoding the transcoded one introduces a gap at the start and makes the segment longer. Given they are both lossy encoding, I would have expected padding to be equally present in both (if at all).
** UPDATE 3 **
According to http://en.wikipedia.org/wiki/Gapless_playback - Only a handful of encoders support gapless - for MP3, I've switched to lame in ffmpeg, and the problem, so far, appears to have gone.
For AAC (see http://en.wikipedia.org/wiki/FAAC), I have tried libfaac (as opposed to libfdk_aac) and it also seems to produce gapless audio. However, the quality of the latter isn't that great and I'd rather use libfdk_aac is possible.
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