Since we now have support for a nice UDP measurement tool in irtt, and also the ability to do fallback runner selection, we finally have an opportunity to retire D-ITG for VoIP measurements.

This issue is a fork of the discussion from the old monster thread in #106 to deal specifically with the VoIP emulation parts. Some comments from that thread:

On Nov 20, 2017, at 10:44 PM, flent-users notifications@github.com wrote:

A goal for me has been to be able to run Opus at 24 bit, 96Khz, with 2.7ms
sampling latency.
Actually getting 8 channels of that through a loaded box would be marvelous.

Sounds like a musician. :) If it were CBR, I don’t know if this is a way to estimate it:

2.7ms ~= 370 packets/sec
@128kbps, 56 bytes / packet (44 data + 12 RTP)
@256kbps, 99 bytes / packet (87 data + 12 RTP)

Just for fun, a ~256 kbps test between two sites, 50km apart, both using p2p WiFi to the Internet. For realtime audio, I guess it’s the maximums that could be the biggest issue.

% ./irtt client -i 2.7ms -l 99 -q -d 10s a.b.c.d

and

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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