The RTP port range that you specify in your config files represent the RTP
ports that will be used at your end... it does not matter what the other
party uses on their end. You can have a wide range or narrow narrow range
depending on the number of simultaneous calls that you expect. In my case
I have 16384–16639 opened up in the firewall (pass EXT-local) and in
rtp.conf. This is to allow inbound media (audio) connections initiated by
the other party. The firewall and the rtp.conf settings need to match. If
they don't then there is a risk that in the SIP conversation to set up a
call your asterisk server tells the remote party to connect to port X and
it will fail if you don't have port X opened in the firewall. Your
asterisk server will always provide X within the range specified in
rtp.conf.
David
On Wed, Sep 4, 2013 at 3:34 AM, 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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