Thanks Lonnie. Oops I didn't look at the config file. So from what you are saying, I don't even need to set these ports if I am setting DSCP correctly. In fact would it be better if I actually left them blank?
I intend on creating QoS Trust boundaries which I can do on the switch. This will rewrite all DSCP to 0 for untrusted hosts (PC's) and the DSCP is trusted for IP Phones. Of course Soft Phones break this concept. So am I correct in saying that EF will go in High Priority and CS3 in medium priority. It says SIP Signalling is default for Medium Priority. Is that classification via Port 5060:5064 on UDP (and TCP/TLS) or via CS3 DSCP classification (or both)? Thanks very much for your help as always. Regards Michael Knill On 18/11/2012, at 1:06 PM, Lonnie Abelbeck wrote: > Michael, > > Personally I use "htb", seems to give me the best results. I think the newer > 'hfsc' might work well for general mixed traffic, but I have found that > 'hfsc' is too slow to adjust for VoIP to be its best. > > Personally I set my Down/Up speed right at my advertised values (28 M / 4 M), > if you want to knock them down by 5% that would be fine, a good starting > point. This will automatically leave about 20% of the up-link free for the > highest priority class. If you need more than 20% for your SIP traffic, then > you can reduce you up-link speed value. > > Regardless you should test by calling yourself via an > inside-to-outside-to-inside call, place yourself on hold, listen to the hold > music. Then run a speed-test, the up-link test will show the worst case > scenario. > > #1 ) In this day and age much of the traffic will be classified by DSCP > markings by default, which is honored, but be sure to set those in sip.conf: > -- > tos_sip=cs3 ; Sets TOS for SIP packets. > tos_audio=ef ; Sets TOS for RTP audio packets. > tos_video=af41 ; Sets TOS for RTP video packets. > tos_text=af41 ; Sets TOS for RTP text packets. > -- > https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service > > Setting the SIP RTP port range is still fine, and what we had to do in the > Asterisk 1.2 days. > > You can also do some tweaks here: > >> From the Network tab -> Firewall Plugins: [ traffic-shaper ] > > You can define/tweak some of the priorities, but the defaults should be good > for the typical situation. Should you have a NAS doing offsite backups or > such, you may want to define SHAPER_P2P_HOSTS. > > > #2) A 5% reduction is fine, you could start with no reduction and start from > there. Each ISP probably has it's own sweet spot. Be sure to test. > > > #3) Downlink shaping is not as critical since the bottleneck is usually on > the up-link, but it simply discards bursts of data higher than the limit. > > It seems to me that getting quality voice connections is easier today than it > was a few years ago. Possibly the bigger pipes are the reason, possibly some > traffic shaping is done by the ISP ? I don't use IAX anymore. :-) Also > people are now so used to cell phones, crappy audio is tolerated more than it > was in the past. :-) > > Regardless enabling traffic shaping is good practice, if for nothing else to > control buffer-bloat. > > Bufferbloat > http://en.wikipedia.org/wiki/Bufferbloat > > Lonnie > > PS: We use 'tc' in the traffic shaper plugin to perform the shaping. > > > On Nov 17, 2012, at 3:53 PM, Michael Knill wrote: > >> To the group >> >> I realise that I don't completely understand how the traffic shaping works >> on AstLinux and I really need to. >> I have set the following: >> >> Type - I am using hfsc which I assume is the best >> Downlink speed - I am setting this to slightly less than than my DSL modems >> reported speed e.g. 95% >> Uplink speed - I am setting this to slightly less than than my DSL modems >> reported speed e.g. 95% >> VoIP UDP Ports - 16384:16639. The same as I have set in rtp.conf >> >> My questions are: >> >> 1) Does the script only use the VoIP ports for queue classification. It >> appears that even though I have the range set in rtp.conf, it only affects >> my allocated port and not the destination port to my Service provider which >> is not in this range? Wont this mean that I am not prioritising rtp on my >> uplink? Can I force symmetrical RTP port allocation? >> 2) Is 95% ok for the reduction. Does anyone use a different formula >> 3) How does the Downlink traffic shaping work? Does it just restrict the >> traffic flow of non VoIP traffic (e.g. not in the VoIP UDP Port Range) to >> fit the Downlink traffic shaping envelope? >> >> What is everyone else doing? >> >> Regards >> Michael Knill > > > > > ------------------------------------------------------------------------------ > Monitor your physical, virtual and cloud infrastructure from a single > web console. 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