[FFmpeg-trac] #6625(undetermined:new): "Freezes" transcoding RTP g.711 stream to mp3

FFmpeg trac at avcodec.org
Wed Aug 30 16:08:42 EEST 2017

#6625: "Freezes" transcoding RTP g.711 stream to mp3
             Reporter:  sagonzal     |                    Owner:
                 Type:  defect       |                   Status:  new
             Priority:  normal       |                Component:
              Version:  unspecified  |  undetermined
             Keywords:  rtp          |               Resolution:
             Blocking:               |               Blocked By:
Analyzed by developer:  0            |  Reproduced by developer:  0

Comment (by sagonzal):

 Truthfully, I’m not entirely certain. My command used to just be:

 ffmpeg -re -f mulaw -i rtp:// -acodec libmp3lame live-
 rtp-g711-toMP3.mp3 -report

 only it didn't work with live intercom audio (0 packets read/encoded,
 etc.), and with the Wireshark audio resulted in an mp3 that sounded very
 high-pitched and fast. After adding sample_rate and asetrate of 9000, the
 mp3 sounded "normal". I figured the same could be applied to the intercom

 I should note that when I spoke with the company that makes the intercoms,
 I was told that the g.711 audio stream would send as either '''7kHz''' or,
 more likely, as '''3.4kHz'''. I haven’t gotten those rates to work for me
 with either rtp or .raw input.

Ticket URL: <https://trac.ffmpeg.org/ticket/6625#comment:2>
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