* [Flent-users] [tohojo/flent] Using irtt for VoIP tests (#119)
@ 2017-11-22 12:09 Toke Høiland-Jørgensen
2017-11-23 16:48 ` Toke Høiland-Jørgensen
0 siblings, 1 reply; 2+ messages in thread
From: Toke Høiland-Jørgensen @ 2017-11-22 12:09 UTC (permalink / raw)
To: tohojo/flent; +Cc: Subscribed
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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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* Re: [Flent-users] [tohojo/flent] Using irtt for VoIP tests (#119)
2017-11-22 12:09 [Flent-users] [tohojo/flent] Using irtt for VoIP tests (#119) Toke Høiland-Jørgensen
@ 2017-11-23 16:48 ` Toke Høiland-Jørgensen
0 siblings, 0 replies; 2+ messages in thread
From: Toke Høiland-Jørgensen @ 2017-11-23 16:48 UTC (permalink / raw)
To: tohojo/flent; +Cc: Subscribed
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Closed #119 via 9d96cbfe0ae6ae9ac1bab13dabe081298fb0ed36.
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