[FFmpeg-user] G711 streaming

[Roger Pack]

On 12/18/12, SCHILL Rolf <ROLF.SCHILL at thalesgroup.com> wrote:
> Hello (ffmpeg) world
>
> I try to stream a wav file to an old fashioned voip device. It provides no
> rtcp or rstp support. It needs the packet with g711 codec content  in 20 ms
> timing.
>
> I try it with
>
> ffmpeg -I nput.wav -re -acodec pcm_alaw -f rtp
> rtp://ip-addr:portnumber?pkt_size=172
>
>
> This command streams well but all packets are send out in more or less a
> half second until the next packets come.

Full uncut command line and console output please?
Maybe it is reading the ".wav" file in 64K chunks, and, since you
specified "-re" it basically sends that 64K worth out, then waits
until a specified time has past, then send the next, etc.
Maybe the asetnsamples filter would help slit it up, or...maybe it is
impossible to get more than 0.5 second granularity here? Dunno.
-r