Are you referring to the "VoIP UDP Ports:" ? The default is fine, it does no harm if set to the 16384:16639 range.
Lonnie On Sep 4, 2013, at 4:57 PM, Michael Knill wrote: > Thanks Lonnie. Yes I am reading about hfsc now and I will do some testing > today. Hopefully they have not fixed the problem yet so I can test! > > Lonnie are you able to comment on the configuration of media UDP ports rules > on the firewall? I have removed all the rules as they dont seem to be > relevant as they dont match the service providers port range anyway? > > 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 disappointin g > . >>> >>>> 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 >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Learn the latest--Visual Studio 2012, SharePoint 2013, SQL 2012, more! >>>> Discover the easy way to master current and previous Microsoft technologies >>>> and advance your career. Get an incredible 1,500+ hours of step-by-step >>>> tutorial videos with LearnDevNow. Subscribe today and save! >>>> http://pubads.g.doubleclick.net/gampad/clk?id=58040911&iu=/4140/ostg.clktrk >>>> _______________________________________________ >>>> 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]. >>> >>> >>> ------------------------------------------------------------------------------ >>> Learn the latest--Visual Studio 2012, SharePoint 2013, SQL 2012, more! >>> Discover the easy way to master current and previous Microsoft technologies >>> and advance your career. Get an incredible 1,500+ hours of step-by-step >>> tutorial videos with LearnDevNow. Subscribe today and save! >>> http://pubads.g.doubleclick.net/gampad/clk?id=58041391&iu=/4140/ostg.clktrk >>> _______________________________________________ >>> 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]. >>> >>> >> >> >> ------------------------------------------------------------------------------ >> Learn the latest--Visual Studio 2012, SharePoint 2013, SQL 2012, more! >> Discover the easy way to master current and previous Microsoft technologies >> and advance your career. Get an incredible 1,500+ hours of step-by-step >> tutorial videos with LearnDevNow. Subscribe today and save! >> http://pubads.g.doubleclick.net/gampad/clk?id=58041391&iu=/4140/ostg.clktrk >> _______________________________________________ >> 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]. > > > ------------------------------------------------------------------------------ > Learn the latest--Visual Studio 2012, SharePoint 2013, SQL 2012, more! > Discover the easy way to master current and previous Microsoft technologies > and advance your career. Get an incredible 1,500+ hours of step-by-step > tutorial videos with LearnDevNow. Subscribe today and save! > http://pubads.g.doubleclick.net/gampad/clk?id=58041391&iu=/4140/ostg.clktrk > _______________________________________________ > 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]. > > ------------------------------------------------------------------------------ Learn the latest--Visual Studio 2012, SharePoint 2013, SQL 2012, more! Discover the easy way to master current and previous Microsoft technologies and advance your career. Get an incredible 1,500+ hours of step-by-step tutorial videos with LearnDevNow. Subscribe today and save! http://pubads.g.doubleclick.net/gampad/clk?id=58041391&iu=/4140/ostg.clktrk _______________________________________________ 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].
