RTP g.711 stream to MP3

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RTP g.711 stream to MP3

Sara Gonzalez

Hi Everyone,

I am trying to connect to a live RTP stream of g.711 audio coming from an intercom and then save it as an mp3 file.
This is the FFmpeg[1] command I have been using:

-re -f mulaw -i rtp://10.200.1.14:32760 -sample_rate 44100 -filter:a "asetrate=9000" -acodec libmp3lame -ar 44100 output.mp3 -report

From the report[2] you can see that the command line outputs some information about opening the input file then “freezes” after
“[udp @ 0000002c748241e0] end receive buffer size reported is 65536”.
The command line will stay that way indefinitely until I enter ctrl+c (q doesn’t work), after which FFmpeg says there were 0 bytes read/encoded, and the output file produced is empty/unplayable[2].

Using Wireshark, I have confirmed that there is traffic coming through port 32760 — packets using the RTP protocol and g.711U payload type, as expected. I right-clicked on one, did “Decode As” RTP, then Analyzed and Saved the data as a "synchronized forward stream audio” in the .raw format[3]. Using the above command with the audio file as the input instead of an RTP stream, I am able to output an mp3 that sounds pretty good.

I’m very new to FFmpeg/working with audio streams and I don’t know why the command works with the saved audio from Wireshark but not with the live stream. I also used gdb [4] and a debug version of FFmpeg [5] to provide a backtrace [6] of the command in case the output is helpful to anyone.

I would appreciate any light that can be shed on this problem!


Thanks very much,

Sara


[1] ffmpeg version "N-86996-g931c0ac95c-Reino" cross-compiled for Windows 64 w/ pthreads and debug=3 enabled (can share file/give more info if needed)
[2] “live-rtp-g711-toMP3-REPORT.log” & “live-rtp-g711-toMP3.mp3” (attached)
[3] I can supply the .raw audio from Wireshark if needed
[4] gdb version 7.1.9 for Windows 32 & 64  (can share file/give more info if needed)
[5] ffmpeg_g version "N-86996-g931c0ac95c-Reino" cross-compiled for Windows 64 w/ pthreads and debug=3 enabled (can share file/give more info if needed)
[6] “live-rtp-g711-toMP3-BACKTRACE.txt” (attached)




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