From: Joerg Dorchain <joerg@dorchain.net>
To: "John A. Sullivan III" <jsullivan@opensourcedevel.com>
Cc: netfilter@vger.kernel.org
Subject: Re: SIP conntrack defeating Asterisk canreinvite
Date: Wed, 26 Aug 2009 09:15:10 +0200 [thread overview]
Message-ID: <20090826071510.GZ6724@Redstar.dorchain.net> (raw)
In-Reply-To: <1251248668.6450.59.camel@jaspav.missionsit.net.missionsit.net>
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On Tue, Aug 25, 2009 at 09:04:28PM -0400, John A. Sullivan III wrote:
> The reinvite works by the Asterisk server sending a SIP invite after the
> call has been set up. The new invite contains the address of the phone
> in the SDP portion of the packet rather than the address of the PBX.
> This should redirect the media stream to flow directly between the
> phones. However, it appears conntrack is rewriting the SDP so that the
> address is reverted to the PBX address.
Rewriting sounds like nat. I am using conntrack_sip to be able
to have the rtp connections accepted as related to a sip
connection. Are you sure that you aren't using the sip nat helper
by change?
To have reinvites working, I needed sip_direct_media=0 as option
to nf_conntrack_cip
Bye,
Joerg
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next prev parent reply other threads:[~2009-08-26 7:15 UTC|newest]
Thread overview: 5+ messages / expand[flat|nested] mbox.gz Atom feed top
2009-08-26 1:04 SIP conntrack defeating Asterisk canreinvite John A. Sullivan III
2009-08-26 7:15 ` Joerg Dorchain [this message]
2009-08-26 10:59 ` John A. Sullivan III
2009-08-27 23:01 ` John A. Sullivan III
2009-09-01 3:11 ` John A. Sullivan III
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