Samples per pkt

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I've been trying to stream audio from ffmpeg into a pjsip stream created
without using SIP.
Having matched up the codecs at both ends I have a problem with inbound
audio, outbound seems fine.
Using pcmu codec I can use the ffmpeg options asetnsamples=b=160 and
aresample=8000 and then inbound audio is ok.
These seem to be logical values.
Using L16/44100/1 codec I have to use aresample=44100 and
asetnsamples=n=441 to get the audio (mono) to be usable.
I would expect the 441 to be 882 samples for this.
Does anyone have a suggestion as to why the sample rate setting should be
half that which logic would suggest and
also why I have to set the sample/pkt in the first place? It is only
inbound audio to pjsip that exhibits the problem, presumably
ffplay as a receiver can cope with pjsip's output.

Any suggestions gratefully received.

PW.

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