[Bug 220078] [patch] [panic] [ipfw] repeatable kernel panic due to unlocked INADDR_TO_IFP usage

2017-09-26 Thread bugzilla-noreply 
https://bugs.freebsd.org/bugzilla/show_bug.cgi?id=220078

--- Comment #20 from commit-h...@freebsd.org ---
A commit references this bug:

Author: ae
Date: Wed Sep 27 01:47:54 UTC 2017
New revision: 324047
URL: https://svnweb.freebsd.org/changeset/base/324047

Log:
  MFC r323839:
Use in_localip() function instead of unlocked access to addresses hash
to determine that an address is our local.

PR: 220078

Changes:
_U  stable/11/
  stable/11/sys/netpfil/ipfw/ip_fw2.c

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Re: FreeBSD, IPFW and the SIP/VoIP NAT problem

2017-09-26 Thread Graham Menhennitt 

On 26/09/2017 10:35 PM, O. Hartmann wrote:

Hello,

trying to build a FreeBSD based router/PBX (Asterisk 13) appliance, I ran into
several problems. My questions might have a "noobish" character, so my apology,
my experiences with IPFW are not as thorough as they should be.

...



The FreeBSD system acting as a router is supposed to have a jail soon
containing the Asterisk 13 IP PBX (at the moment running on the main system).
To provide access to the VoIP infrastructure inside/behind the router/NAT
system, the in-kernel NAT facility of FreeBSD is forwarding the relevant
UPD/TCP ports for VoIP to its destination network, and here I have a problem to
solve.


Does your VoIP provider allow IAX2 protocol as well as SIP? Many do. 
Then you can avoid the problem completely.


Graham
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FreeBSD, IPFW and the SIP/VoIP NAT problem

2017-09-26 Thread O. Hartmann 
Hello,

trying to build a FreeBSD based router/PBX (Asterisk 13) appliance, I ran into
several problems. My questions might have a "noobish" character, so my apology,
my experiences with IPFW are not as thorough as they should be.

Before I'll got into medias res, aquestion about Pine64/AARCH64:

- port net/asterisk13 is supposed to build only on armv6, is aarch64 about
  coming soon also supported? 
- would a Pine64 running CURRENT (12) be sufficient as a PBX platform (assumed
  having 2 GB of RAM)?

My main concern is about IPFW (we do not use PF for some reasons, I have to
stay with IPFW).

I'm a customer of two ITSPs and my SoHo network is behind NAT and not yet IPv6.
The FreeBSD system acting as a router is supposed to have a jail soon
containing the Asterisk 13 IP PBX (at the moment running on the main system).
To provide access to the VoIP infrastructure inside/behind the router/NAT
system, the in-kernel NAT facility of FreeBSD is forwarding the relevant
UPD/TCP ports for VoIP to its destination network, and here I have a problem to
solve.

While it is sumple and easy to forward 5060/udp, 5070/tcp and other ports, it
is a mess and pain in the arse to forward a whole range, say 11000/udp -
35000/udp, which is a range one of my providers is sending RTP on. A second
provider uses another range for RTP, starting at 5000/udp. So, the logical
consequence would be a union set up UDP range to forward, which exapnds then
form 5000/udp to 45000/udp - which is much more a pain ...

One of the most disturbing and well known problems is that due to the stateful
firewall the RTP session very often is half duplex - it seems one direction
of the RTP connection doesn't make it through IPFW/NAT. As often I search the
net, I always get informed this is a typical problem and solutions are
provided by so called ALGs - since SIP protocol's SDP indicates within the
payload of the packets on which UDP ports both ends wish to establish their
RTP session, it would be "easy" to pinhole the IPFW on exactly those ports for
a theoretical large number of sessions, if IPFW could "divert" those packets
to an instance inspecting SDP (or whatever is used for the RTP port
indication, I'm new to that, sorry for the terminology) and then pinholing the
NAT/IPFW for exactly this purpose without the forwarding mess. I came along
netgraph() while searching for hints and hooks, but it seems a complete Linux
domain, when it somes to appliances like VoIP/IP PBX.

Either, the problem is that trivial on FreeBSD, so no further mentioning is
necessary (which would explain the vast emptyness of explanations, hints and
so on) or FreeBSD is a complete wasteland on this subject - which I also
suspect, since pfSense and OPNsense must have come along with such problems
and I simply do not know or recognise the software used for those purposes.

So, if someone enlightened in this matter stumbles over my question and could
delegate me onto the right way (ports, ng_XXX netgraph ficilities to look at,
some ipfw techniques relevant to the problem apart from the stupid simple
forwarding large ranges of ports) - I'd appreciate this and

thanks in advance for patience and help,

Oliver 
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Re: FreeBSD, IPFW and the SIP/VoIP NAT problem

2017-09-26 Thread Damjan Jovanovic 
On Tue, Sep 26, 2017 at 3:44 PM, O. Hartmann 
wrote:

> On Tue, 26 Sep 2017 11:00:45 +0200
> Damjan Jovanovic  wrote:
>
> > On Tue, Sep 26, 2017 at 10:35 AM, O. Hartmann 
> > wrote:
> >
> > > Hello,
> > >
> > > trying to build a FreeBSD based router/PBX (Asterisk 13) appliance, I
> ran
> > > into
> > > several problems. My questions might have a "noobish" character, so my
> > > apology,
> > > my experiences with IPFW are not as thorough as they should be.
> > >
> > > Before I'll got into medias res, aquestion about Pine64/AARCH64:
> > >
> > > - port net/asterisk13 is supposed to build only on armv6, is aarch64
> about
> > >   coming soon also supported?
> > > - would a Pine64 running CURRENT (12) be sufficient as a PBX platform
> > > (assumed
> > >   having 2 GB of RAM)?
> > >
> > > My main concern is about IPFW (we do not use PF for some reasons, I
> have to
> > > stay with IPFW).
> > >
> > > I'm a customer of two ITSPs and my SoHo network is behind NAT and not
> yet
> > > IPv6.
> > > The FreeBSD system acting as a router is supposed to have a jail soon
> > > containing the Asterisk 13 IP PBX (at the moment running on the main
> > > system).
> > > To provide access to the VoIP infrastructure inside/behind the
> router/NAT
> > > system, the in-kernel NAT facility of FreeBSD is forwarding the
> relevant
> > > UPD/TCP ports for VoIP to its destination network, and here I have a
> > > problem to
> > > solve.
> > >
> > > While it is sumple and easy to forward 5060/udp, 5070/tcp and other
> ports,
> > > it
> > > is a mess and pain in the arse to forward a whole range, say 11000/udp
> -
> > > 35000/udp, which is a range one of my providers is sending RTP on. A
> second
> > > provider uses another range for RTP, starting at 5000/udp. So, the
> logical
> > > consequence would be a union set up UDP range to forward, which exapnds
> > > then
> > > form 5000/udp to 45000/udp - which is much more a pain ...
> > >
> > > One of the most disturbing and well known problems is that due to the
> > > stateful
> > > firewall the RTP session very often is half duplex - it seems one
> direction
> > > of the RTP connection doesn't make it through IPFW/NAT. As often I
> search
> > > the
> > > net, I always get informed this is a typical problem and solutions are
> > > provided by so called ALGs - since SIP protocol's SDP indicates within
> the
> > > payload of the packets on which UDP ports both ends wish to establish
> their
> > > RTP session, it would be "easy" to pinhole the IPFW on exactly those
> ports
> > > for
> > > a theoretical large number of sessions, if IPFW could "divert" those
> > > packets
> > > to an instance inspecting SDP (or whatever is used for the RTP port
> > > indication, I'm new to that, sorry for the terminology) and then
> pinholing
> > > the
> > > NAT/IPFW for exactly this purpose without the forwarding mess. I came
> along
> > > netgraph() while searching for hints and hooks, but it seems a complete
> > > Linux
> > > domain, when it somes to appliances like VoIP/IP PBX.
> > >
> > > Either, the problem is that trivial on FreeBSD, so no further
> mentioning is
> > > necessary (which would explain the vast emptyness of explanations,
> hints
> > > and
> > > so on) or FreeBSD is a complete wasteland on this subject - which I
> also
> > > suspect, since pfSense and OPNsense must have come along with such
> problems
> > > and I simply do not know or recognise the software used for those
> purposes.
> > >
> > > So, if someone enlightened in this matter stumbles over my question and
> > > could
> > > delegate me onto the right way (ports, ng_XXX netgraph ficilities to
> look
> > > at,
> > > some ipfw techniques relevant to the problem apart from the stupid
> simple
> > > forwarding large ranges of ports) - I'd appreciate this and
> > >
> > > thanks in advance for patience and help,
> > >
> > > Oliver
> > >
> >
> >
> > Hi
> >
> > It might be easier if you just enable STUN on Asterisk, and build FreeBSD
> > from source with my [largely neglected :( ] patch on
> > https://bugs.freebsd.org/bugzilla/show_bug.cgi?id=219918
> >
> > That way Asterisk should dynamically discover consistent external
> mappings
> > for connections, making port forwarding RTP unnecessary.
> >
> > Damjan
>
> STUN is enabled, but my providers do not support STUN.
>
> I try to figure out how SIP works exactly to make my problem more precise.
> I
> also try to understand the aim of your patch - as far as I know, it does
> exactly as it is needed for the IPW/NAT/VoIP case. And I really regret that
> there are objections to commit the patch ...
>
>
Firstly, if your providers support NAT, you register to them (as opposed to
they register to you, or no registration), and the only VoIP calls are
to/from your providers and to/from the same IP:port you register to (as
opposed to unknown external addresses), then none of this should be
necessary. Just put these on every SIP peer

[Bug 220078] [patch] [panic] [ipfw] repeatable kernel panic due to unlocked INADDR_TO_IFP usage

2017-09-26 Thread bugzilla-noreply 
https://bugs.freebsd.org/bugzilla/show_bug.cgi?id=220078

Kubilay Kocak  changed:

   What|Removed |Added

  Flags||mfc-stable11+

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