Michael,

That is not a simple question, the 'tc' command is a very complicated command.

The traffic-shaper plugin is in the upstream Arno's iptables firewall, but we 
have our own version with a minor addition here:
http://sourceforge.net/p/astlinux/code/HEAD/tree/branches/1.0/package/arnofw/traffic-shaper/60traffic-shaper.plugin.sh

Hopefully following the code will help a little.

The OpenWRT docs has some info "Network Traffic Control"
http://wiki.openwrt.org/doc/howto/packet.scheduler/packet.scheduler

>From the CLI, this will output tc and iptables statistics, understanding the 
>tc stuff may not be obvious.

$ arno-iptables-firewall status-plugins traffic-shaper


Lonnie


On Sep 4, 2013, at 6:56 PM, Michael Knill wrote:

> There are probably some variables that are not known but I dont think it 
> should be too black as such.
> Safest thing to do if you can is htb with plenty of headroom for voice.
> 
> One thing I dont quite understand is what is actually shaped? The ports you 
> have in traffic-shaper.conf? How does the low, medium and high priority work? 
> If its DSCP EF does it just bypass the shaper with htb? How does it work?
> 
> Also are there any commands can I use to show what configuration is running, 
> statistics etc?
> 
> Regards
> Michael Knill
> 
> 
> 
> 
> On 05/09/2013, at 9:01 AM, Lonnie Abelbeck <[email protected]> wrote:
> 
>> Michael,
>> 
>> I have found traffic shaping to be somewhat of a black art.  Making multiple 
>> outbound calls while doing an outbound speed-test (up and down) is a good 
>> way to test.
>> 
>> I suspect your DSL link may behave differently than my cable modem 
>> connection.
>> 
>> Lonnie
>> 
>> 
>> On Sep 4, 2013, at 5:42 PM, Michael Knill wrote:
>> 
>>> Ok after my testing hfsc does indeed work this way. Although not perfect, 
>>> it was more than acceptable during the rare periods of extreme congestion. 
>>> I noticed that the effective rate did not change when calls were introduced 
>>> (I tried 3). This makes me happy :)
>>> 
>>> So this is how I think it works out:
>>> 
>>> If you have a shared broadband service with plenty of upload headroom, you 
>>> should use htb and your shaping parameter should be (Upload speed + 25%) - 
>>> (100K (G711) * max number of channels) e.g. Upload speed 1000K with 4 
>>> channels required = 1000 + 250 - 400 = 850K.
>>> 
>>> If you dont have much upload headroom and it will be a problem to shape it 
>>> right down, then just configure for hfsc at Upload Speed - 10%-20% to be 
>>> conservative.
>>> 
>>> What do you think?
>>> 
>>> Regards
>>> Michael Knill
>>> 
>>> 
>>> 
>>> 
>>> On 05/09/2013, at 7:46 AM, Lonnie Abelbeck <[email protected]> 
>>> wrote:
>>> 
>>>> Michael,
>>>> 
>>>> AstLinux's traffic shaper honors QoS values using DSCP matching.  Of 
>>>> course you have to enable this in Asterisk sip.conf:
>>>> --
>>>> ; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for 
>>>> a description of these parameters.
>>>> 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.
>>>> --
>>>> 
>>>> The traffic shaper honors "htb" and "hfsc" methods, the very old "htb" 
>>>> works as I described and is what I use on my 28/4 Mbps connection.  The 
>>>> "hfsc" method is newer and works more like what you want by dynamically 
>>>> prioritizing, but in my tests (years ago) the dynamic slew was too slow 
>>>> for perfect voice in worst case conditions.  Feel free to try "hfsc".
>>>> 
>>>> Again, I suggest you disable downlink shaping.
>>>> 
>>>> Keep in mind the traffic shaper is implemented in the Linux kernel and the 
>>>> 'tc' command.  We can't make it any better than how it is implemented in 
>>>> Linux.
>>>> 
>>>> As for "the VoIP UDP Ports range in the traffic shaper config" it does not 
>>>> matter much anymore if you enable QoS marking in Asterisk, mostly a legacy 
>>>> setting. A DSCP match is the best way to go when possible, and automatic.
>>>> 
>>>> Lonnie
>>>> 
>>>> 
>>>> 
>>>> On Sep 4, 2013, at 4:10 PM, Michael Knill wrote:
>>>> 
>>>>> Ok well the traffic shaping does not work the way I thought then which is 
>>>>> a big problem as far as I am concerned. So Lonnie what you are saying is 
>>>>> that you essentially have to configure the traffic shaper to allow the 
>>>>> headroom for the maximum number of supported channels? This basically 
>>>>> takes the system back to the multiplexor days. The way that QoS and 
>>>>> traffic shaping should work is that the full shaped envelope has 
>>>>> prioritised traffic with it so that the full bandwidth can be used at all 
>>>>> times by all traffic types and bandwidth restrictions only apply during 
>>>>> congestion. This is how a Cisco router works and I thought this worked 
>>>>> the same way. This is really disappointing to find this out now as it was 
>>>>> a significant decider in me using Astlinux. I am going to need to use a 
>>>>> separate broadband connection for voice for most businesses as they will 
>>>>> not tolerate the significant reduction of upload speed or I will need to 
>>>>> use a true QoS capable router which would be really disappoint
 i
> ng
>>> .
>>>>> 
>>>>>> From my perspective, the firewall, routing and QoS areas of Astlinux are 
>>>>>> the areas that I would like to have the most development as this is what 
>>>>>> differentiates Astlinux from nearly all other distributions, not the 
>>>>>> fact that it runs Asterisk.
>>>>> 
>>>>> Getting back to the RTP ports, I still dont know what the VoIP UDP Ports 
>>>>> range does in the traffic shaper config. Can someone please tell me?
>>>>> I have set 16384:16639 as recommended. Now after doing an RTP Debug from 
>>>>> making an outgoing external phone call, this is what I get:
>>>>> 
>>>>> Sent RTP P2P packet to 203.2.134.1:21294 (type 08, len 000160)
>>>>> Sent RTP P2P packet to 172.30.30.116:16458 (type 08, len 000160)
>>>>> Sent RTP P2P packet to 203.2.134.1:21294 (type 08, len 000160)
>>>>> Sent RTP P2P packet to 172.30.30.116:16458 (type 08, len 000160)
>>>>> Sent RTP P2P packet to 203.2.134.1:21294 (type 08, len 000160)
>>>>> Sent RTP P2P packet to 172.30.30.116:16458 (type 08, len 000160)
>>>>> 
>>>>> Yes the port is correct for the internal IP Phone (172.30.30.116) but the 
>>>>> external provider port is negotiated by them (203.2.134.1). So what is 
>>>>> the point of putting 16384:16639 in the shaper and the firewall rules. 
>>>>> Its doesn't even see these ports as they are sent back to the internal 
>>>>> interface?
>>>>> 
>>>>> I would really like to get this sorted as it significantly affects my 
>>>>> whole system architecture and at this stage, for sites that share a 
>>>>> single broadband connection, I will have to use a separate QoS capable 
>>>>> router.
>>>>> 
>>>>> Regards
>>>>> Michael Knill
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> On 05/09/2013, at 1:05 AM, Lonnie Abelbeck <[email protected]> 
>>>>> wrote:
>>>>> 
>>>>>> Hi Michael,
>>>>>> 
>>>>>> First, I would disable Downlink shaping, the latest version of AstLInux 
>>>>>> have a "Disabled" option, with your 1.0.4 version setting "Downlink 
>>>>>> Speed" to "0" should work IIRC, though official in AstLinux 1.1.0 .
>>>>>> 
>>>>>> The "Uplink Speed" reserves about 25% for VoIP traffic, so a value of 
>>>>>> 1000 Kbps will limit low priority traffic to about 750 Kbps, leaving 250 
>>>>>> Kbps for VoIP if the real limit is 1000 Kbps.  This would support 2 SIP 
>>>>>> calls using a ulaw codec.
>>>>>> 
>>>>>> Keep lowering "Uplink Speed" to provide more headroom for VoIP calls.
>>>>>> 
>>>>>> As far as the VoIP UDP port range, the range should match your Asterisk 
>>>>>> rtp.conf settings (reduce the default) and match a manual allowed UDP 
>>>>>> port range in your firewall (Pass EXT->Local).  Do not enable the 
>>>>>> sip-voip plugin.
>>>>>> 
>>>>>> In practice any *phone* that registers with your Asterisk should have 
>>>>>> the same VoIP UDP port range defined in their configs.  A range (rtpend 
>>>>>> - rtpstart) of 256 ports should be fine, don't make it too large.
>>>>>> 
>>>>>> Lonnie
>>>>>> 
>>>>>> 
>>>>>> On Sep 4, 2013, at 7:22 AM, Michael Knill wrote:
>>>>>> 
>>>>>>> I have been doing some more testing with iftop. My ADSL service is 
>>>>>>> clocking at 947K Up so I have set  900K as the upload speed.
>>>>>>> iftop is showing a constant 800K or so peak ppp0 interface transfer 
>>>>>>> rate. I assume this is not 900 due to overheads etc. This rate does not 
>>>>>>> change much when a call is included into the traffic. Although the data 
>>>>>>> traffic reduces, there is considerable voice packet loss.
>>>>>>> If I change the shape rate to 800, my ppp0 interface transfer rate 
>>>>>>> reduces to around 700K as expected. Now when I add a voice call, there 
>>>>>>> is no packet loss, however the interface traffic rate goes to about 
>>>>>>> 780K which indicates that it is not being included in the shaped 
>>>>>>> envelope. When I add a second call, I get packet loss again as it is 
>>>>>>> taken past the maximum upload rate.
>>>>>>> 
>>>>>>> To me this indicates that the voice traffic is not being included in 
>>>>>>> traffic shaping or traffic shaping is not even working. I turned it off 
>>>>>>> and it really didn't make any difference.
>>>>>>> The box is using version 1.0.4. Any ideas?
>>>>>>> 
>>>>>>> Regards
>>>>>>> Michael Knill
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> On 04/09/2013, at 5:34 PM, Michael Knill 
>>>>>>> <[email protected]> wrote:
>>>>>>> 
>>>>>>>> To the group
>>>>>>>> 
>>>>>>>> I am still very confused about what I should be setting the VoIP UDP 
>>>>>>>> port range to. I use different providers with different ranges. Do I 
>>>>>>>> just set it to 10000 - 65535? 
>>>>>>>> What does it actually do?
>>>>>>>> 
>>>>>>>> In the Astlinux Firewall Addins doco it says for sip-voip:
>>>>>>>> 
>>>>>>>> This plugin attempts to track the RTP ports used in a SIP dialog and 
>>>>>>>> automatically open the necessary RTP ports when needed.
>>>>>>>> In practice this plugin does not always yield the expected results. 
>>>>>>>> Feel free to experiment.
>>>>>>>> When this plugin is disabled (the default) the SIP RTP ports must be 
>>>>>>>> manually opened to match the Asterisk rtp.conf rtpstart/rtpend values.
>>>>>>>> 
>>>>>>>> The rtpstart and rtpend values I have in rtp.conf are not what my 
>>>>>>>> provider(s) use. Should I change it to match? How come I have no sip 
>>>>>>>> firewall rules as mentioned above but it still works fine?
>>>>>>>> How does the firewall know to open up the media ports? In all the 
>>>>>>>> tests I did, the port was the same so does it just set up a stateful 
>>>>>>>> translation?
>>>>>>>> 
>>>>>>>> This really started with one of my customers today whereby they were 
>>>>>>>> significantly congesting their broadband link (yes working on that) 
>>>>>>>> but their existing telco service was working fine (getting dropouts 
>>>>>>>> but the voice was fine, albeit delayed). I added another service from 
>>>>>>>> another Telco (before I realised it was congested) and they were 
>>>>>>>> having lots of upstream voice quality problems.
>>>>>>>> 
>>>>>>>> Is there anything that could cause one service to be matched in the 
>>>>>>>> traffic shaper and another not?
>>>>>>>> 
>>>>>>>> Regards
>>>>>>>> Michael Knill
>>>>>> 
>>>>>> 
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>>>>> 
>>>>> 
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>> 
>> 
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> 
> 
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