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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