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• Revision 9e23bd5df5 : Merge "Merge branch 'master' into experimental" into experimental

  26 avril 2013, par Johann

  Changed Paths :
   Modify /vp9/decoder/vp9_decodframe.c

  
  
  Merge "Merge branch 'master' into experimental" into experimental

• Using FFMPEG for Bidirectional Voice Communication with Symmetric RTP

  3 février 2024, par Batuhan Öksüz

  I 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.

  


• question regarding ffmpeg webcam stream (RTMP)

  7 janvier 2021, par J Hwang

  I am a novice if you will regarding FFMPEG.
I was tasked with creating a Windows-based C++ project that takes a live webcam stream, does the necessary decoding and encoding, and sends it via RTMP to a server.
The constraints were that the output format be "flv" due to the design of the server end and that it take both audio and video input. Quality of the stream isn't that much of an issue as long as the audio doesn't desync too much from the video.

  


  I drew on the prior works of https://github.com/leixiaohua1020 in simplest_ffmpeg_streamer and simplest_ffmpeg_device.

  


  Specifically, I tried to mesh together the logic behind the readcamera project to provide an input stream and send it to an output address using the streamer part that previously took a local flv video as input.

  


  However, I ran into some problems.

  


  First of all, I know vfwcap is very much outdated but it seems dshow has some problems with the input part as well as the output part so I'm opting for whichever works first.

  


  Second of all, I don't really understand how to set the codec for the input. The input is using rawvideo by default and I tried to add settings as shown.
pFormatCtx->video_codec_id = AV_CODEC_ID_MPEG4;
However, none of the even slightly relevant codec settings seemed to work. All showed the same error message in the vfwcap error message image below.

  


  I would very much appreciate any pointers on achieving 1. change of codec in input stage 2. fixes for the dshow case

  


  I have put the full code up on github with the appropriate mentions.
https://github.com/luorix1/ffmpeg

  


  error message_dshow

  


  error message_vfwcap