On Sun, Jan 29, 2017 at 1:06 PM, Dave Taht <[email protected]> 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 _______________________________________________ Bloat mailing list [email protected] https://lists.bufferbloat.net/listinfo/bloat
