[FFmpeg-user] problem with converting rtp packets to an audio format

M. V. bored_to_death85 at yahoo.com
Tue Jul 30 23:11:29 CEST 2013

I searched more and more but couldn't find what am I doing wrong. so I think I should simply ask this:

- can ffmpeg read rtp (+sip/sdp) packets from a file and converts them to an output audio file? if yes, how?

I know it's a basic question but I really couldn't find the correct answer to that with document reading and googling.

I would be glad if someone gives me a hint or suggestion.
thank you.

> From: M. V. <bored_to_death85 at yahoo.com>
>To: "ffmpeg-user at ffmpeg.org" <ffmpeg-user at ffmpeg.org> 
>Sent: Tuesday, July 30, 2013 1:25 AM
>Subject: [FFmpeg-user] problem with converting rtp packets to an audio format
>hi guys,
>I'm new to ffmpeg. I'm developing a program which receives and recognizes rtp (+sip/sdp) packets from interface and writes rtp content (just rtp part, not udp and ip header) to a file. now I wanna use "ffmpeg"  to convert this file to an audio format. for this I use:
>    #ffmpeg -f rtp -i /locaton/of/my/rtp/file output.au
>and I receive this error:
>[rtp @ 0x2d62880] Guessing on RTP content - if not received properly you need an SDP file describing it
>Guessed Channel Layout for  Input Stream #0.0 : mono
>Input #0, rtp, from '/location/to/my/rtp/file':
>  Duration: N/A, bitrate: 64 kb/s
>    Stream #0:0: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
>Output #0, au, to '/output.au':
>  Metadata:
>    encoder         : Lavf55.12.100
>    Stream #0:0: Audio: pcm_s16be ([3][0][0][0] / 0x0003), 8000 Hz, mono, s16, 128 kb/s
>Stream mapping:
>  Stream #0:0 -> #0:0 (pcm_alaw -> pcm_s16be)/location/to/my/rtp/file: Connection timed out
>size=       0kB time=-577014:-32:-22.-77 bitrate=N/A    
>size=       0kB time=00:00:00.00 bitrate=N/A    
>video:0kB audio:0kB subtitle:0 global headers:0kB muxing overhead inf%
>Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
>so what am I doing wrong?
>- should I write the whole rtp packets (with ethernet/ip/udp headers) to file or just rtp-section of packets are enough?
>- do I also need to write all related sdp (and sip) packets to a separate file?
>- if so, how can I introduce this file to ffmpeg?
>I googled a lot and checked documentation, but didn't find an answer. (most of what I found was about playing rtp from ffmpeg)
>any help would be much appreciated.
>thank you.
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>ffmpeg-user at ffmpeg.org

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