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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