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Valkaama DVD Label
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Mis à jour : Février 2013
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Autres articles (111)
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Script d’installation automatique de MediaSPIP
25 avril 2011, parAfin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
La documentation de l’utilisation du script d’installation (...) -
Que fait exactement ce script ?
18 janvier 2011, parCe script est écrit en bash. Il est donc facilement utilisable sur n’importe quel serveur.
Il n’est compatible qu’avec une liste de distributions précises (voir Liste des distributions compatibles).
Installation de dépendances de MediaSPIP
Son rôle principal est d’installer l’ensemble des dépendances logicielles nécessaires coté serveur à savoir :
Les outils de base pour pouvoir installer le reste des dépendances Les outils de développements : build-essential (via APT depuis les dépôts officiels) ; (...) -
Automated installation script of MediaSPIP
25 avril 2011, parTo overcome the difficulties mainly due to the installation of server side software dependencies, an "all-in-one" installation script written in bash was created to facilitate this step on a server with a compatible Linux distribution.
You must have access to your server via SSH and a root account to use it, which will install the dependencies. Contact your provider if you do not have that.
The documentation of the use of this installation script is available here.
The code of this (...)
Sur d’autres sites (9708)
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Using FFMPEG for Bidirectional Voice Communication with Symmetric RTP
3 février 2024, par Batuhan ÖksüzI am trying to make a voice communication between two peers. The first peer is my local machine which is behind a symmetric NAT. The second peer is my server running on an AWS EC2 device which has a public IP address. I want to use FFMPEG for sending the audio stream through RTP while at the same time listening to a known port to receive the audio stream the remote peer sends to my device. In order to not deal with NAT traversal issues, I want to be able to use the same IP address and port number I use for sending on my local device for receiving. Is this plausible ? I'm starting to think that this is not possible and here is my rationale :


- 

- If my understanding is correct UDP doesn't allow full-duplex communication ; that is, one cannot use one IP:port pair for both sending and receiving data packets at the same time. Is that correct ?
- If one wants to bind a socket that is already been bound, the function throws a bind error saying "Address already in use.". I figured there is a UDP option to allow binding to the same port or address even if they are in use. These options are namely SO_REUSEADDR and SO_REUSEPORT. However, this page states that it is only possible if the port is in TIME_WAIT state. So this also supports my suspicion.






On the other hand, there is symmetric RTP that clearly states a device can receive RTP streams from an address and port that it simultaneously uses to transmit RTP streams from. With exact words of the RFC :




A device supports symmetric RTP if it selects, communicates, and uses
IP addresses and port numbers such that, when receiving a
bidirectional RTP media stream on UDP port "A" and IP address "a", it
also transmits RTP media for that stream from the same source UDP
port "A" and IP address "a". That is, it uses the same UDP port to
transmit and receive one RTP stream.






A device that doesn't support symmetric RTP would transmit RTP from a
different port, or from a different IP address, than the port and IP
address used to receive RTP for that bidirectional media steam.




So this is where I get confused. Is symmetric RTP somehow works around the limitations of UDP ? How am I getting this wrong ?


Now going back to FFMPEG and the use of symmetric RTP, I see there is an rtp option we can use to set it up, the so called localrtpport=n. I can find almost no explanation to what it does and how it's useful though. Can anyone help me with that ? As far as I can tell, this option tells FFMPEG to use port "n" as the outbound port when transmitting an RTP stream. So if the receiver transmitted its stream to this port then the problem of symmetric NAT requirement would be resolved. Or so I thought...


To draw you a better picture, here are my FFMPEG commands (I'm trying everything in my local host in these commands) :


# My Mac with en0 IP of 192.168.1.64
ffmpeg \
-hide_banner \
-re \
-fflags +genpts \
-f lavfi -i aevalsrc="sin(400*2*PI*t)" \
-ar 8000 \
-f mulaw \
-f rtp -reuse 1 "rtp://192.168.1.72:9193?reuse=1&localrtpport=16386&localrtcpport=16387" \
-protocol_whitelist udp,rtp \
-i rtp://192.168.1.64:16386 \
audio_signal_with_symmetric_rtp.mp3



Here I am simply generating a fixed sound signal and outputting it in mulaw format through rtp. I am using the 'localrtport' option to set my outbound port and I am expecting to receive the remote peer's stream on the same port. This command starts running and and waits for the incoming stream. As soon as I start transmitting the stream from my Raspberry Pi which is on the same wireless network, I get the dreaded "Address already in use." error and the process terminates. Here is the command I use on the Raspberry :


# My Raspberry Pi with wlan0 IP of 192.168.1.72
ffmpeg \
-hide_banner \
-re \
-f lavfi -i aevalsrc="sin(400*2*PI*t)" \
-ar 8000 \
-f mulaw \
-f rtp "rtp://192.168.1.64:16386?reuse=1&localrtpport=16386"



TLDR


The short form of the question comes down to this : How can I make use of symmetric RTP with FFMPEG, receive a stream from the same port I am actively transmitting another stream from ? Is what I'm asking impossible ? Should I go for an alternate route and try to set up a TURN server for my system ? Any help would be appreciated.


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Use data to develop impactful video content
28 septembre 2021, par Ben Erskine — Analytics Tips, Plugins -
ffmpeg, live MPEG-TS demux & decode
8 mai 2017, par NadavRubEnvironment
- Ubuntu-14
- C++
- ffmpeg
Use-case
- Live SPTS is received via UDP by a 3rd party module
- TS Packets are received iteratively
- The TS Video (ES) should be decoded in minimal latency
Considered Implementation
- Upon TS packet reception, immediately push it to the TS demux
- Once enough packets are received the video format is resolvable, create the video codec
- Push each video packet into the video decoder
- Once enough video packets were processed the video codec result a valid output frame
Problem at-hand
Can this be done w/ ffmpeg ?!?!, … using “avformat_open_input” mandate a file to read from… I need a way where I can iteratively push packets to the TS demuxer ( w/ minimal latency )…
Does ffmpeg support the above mentioned use-case ? How ?