[Bloat] Cake "out of tree" on Debian

[Kristian Kielhofner]

On Sun, Jan 29, 2017 at 1:06 PM, Dave Taht <dave.taht at gmail.com> wrote:
>
>
> Groovy. I thought you were "Mr. VOIP"?
>
> While the default fq scheme works really well in cake, if you test
> marking packets as voip (EF, CS4,CS6,CS7,VA), it will end up in the
> diffserv3 voice queue.
>
> asterisk used to have an encapsulating protocol called iax2, which
> generated a single flow as backhaul - is that still deployed?
>
> We have a tool in flent based on dit-itg to test this. It's a bit
> painful to setup the first time. I've longed to have a full
> asterisk/freeswitch/jitsy test setup to look harder at voip/video
> characteristics.
>
> There are also now several as yet underdocumented options in cake:
>
> "nat" will look at flows before they are natted so as to isolate them better.
> When combined with the dual-dsthost or dual-srchost option (depending
> on traffic direction) that gives you per host fq, along with per-flow
> fq.
>
> (nat triple-isolate should also do this but we're still sorting out a
> bug on that:
> https://github.com/dtaht/sch_cake/issues/46
> )
>
> wash: washes out dscp markings. Helpful when your provider (:cough:
> comcast) remarks nearly all traffic to CS1.
>
> I just found that appear.in is using the new "goog" marker, which
> marks all videoconferencing traffic as AF41, which is more or less
> appropriately handled in the "diffserv4" model.
>
> In general I have always had good results with the simplest
> (besteffort or diffserv3) settings.
>
> There's also new support for a docsis mode using the new "mpu" idea.
>
> https://github.com/dtaht/sch_cake/pull/45
>

Thank you but I'm not sure I was ever "Mr. VOIP"!

Regarding what I would broadly call "QoS" with other priorities and
projects my personal experience looks something like this:

WonderShaper ----------> CoDel/SQM in OpenWRT "just works" ---> "Cake
looks really cool" (rabbit hole...)

Asterisk still has iax2 although I haven't used it in at least 12
years. If I were to guess iax2 probably represents a tiny fraction of
what I'd call "realtime/VoIP traffic" in networks. However, it's
probably overly represented in smaller deployments, hobbyists, and
home users.

In 2017 I'd suggest looking closer at a FreeSWITCH test setup:

- WebRTC (as well as "traditional" SIP/RTP, of course)
- One of the most robust OPUS implementations I'm aware of; with
dynamic and adaptive FEC (forward error correction) and PLC (packet
loss concealment)
- Relatively rich (in the open source ecosystem) video support
(codecs, "MCU functionality", etc):

https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video

-- 
Kristian Kielhofner