G.711 can be simulated today with `-i 20ms -l 172 -fill rand -fillall`. I do
this test pretty often, and I think it would be a good default voip test. The
reason for the 172 vs 160 is the addition of a 12 byte RTP header, which is
present in the wireshark trace of a SIP G.711 call:
https://wiki.wireshark.org/SampleCaptures?action=AttachFile&do=get&target=SIP_CALL_RTP_G711
GSM is older now and I'm not sure how much it's still used over the Internet,
but since it has a payload size of 33 bytes(?), some statistics would have to
be sacrificed. I'd give up server received stats and dual timestamps, so `-i
20ms -l 33 -rs none -ts midpoint` is a start. Not sure about additional headers.
It should possible to simulate Opus in CBR mode in a similar way. But Opus also
supports VBR, which would require varying packet sizes, which irtt can't yet do
(plus, this would invalidate or at least pollute the IPDV calculation).
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