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. Get in-depth insight into apps, servers, databases, vmware, SAP, cloud infrastructure, etc. Download 30-day Free Trial. Pricing starts from $795 for 25 servers or applications! http://p.sf.net/sfu/zoho_dev2dev_nov _______________________________________________ Astlinux-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [email protected].
