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].
