No, it is not call stateful. I still keep all the messages regarding
loose_route on my todo list for review, because there were many of them
and I am not sure where does the confusion come from. I'll get back to
it.
Jan.
On 06-05-2005 11:20, Martin Koenig wrote:
reticent wrote:
Currently (in rel_0_9_) the
"loose_route()" function will NOT match a
message unless SER has some stateful information regarding that
transaction in memory (my assumption has been that the information is
created when the "record_route()" or "record_route_preset()"
functions
are used and is referenced by "loose_route()" according to the
"tag=..."
header field inside the "Route:" header), so simply adding a "Route:
..." header into the sip message will not be loose_routed.
Could a Developer please ACK / non-ACK this that Ser suddenly turned
call stateful?
Regards,
Martin
Michael Ulitskiy wrote:
Hello,
I'm trying to comprehend loose routing concept and I have
a question that concerns me.
As far as I understand loose routing says that if there're Route
headers in a message it should be forwarded according to the URIs
set in Route headers.
I thought that this is true only within a dialog, but RFC3261 (part 16.6)
says:
"Requests establishing a dialog may contain a preloaded Route header
field."
Also SER manual says: " the failure not to include loose routing in your
scripts may lead to infinite loops. Make sure that you include the
following script fragment immediately after request sanity checks" and
provide the following
piece of code:
if (loose_route()) {
t_relay();
break;
};
which as far as I understand unconditionally forwards message if Route
header
is present.
So I'm wondering what about security? If I follow this guidelines how I
would
shield my PSTN gateway if anyone can construct message and
pre-load it with URI of my gateway and all my proxies must honor it.
For example I have a PSTN gateway on ip address 10.1.1.5 and proxy
on 10.1.1.10 that supposed to interface outside world.
So I guess if someone construct a message like this:
INVITE sip:12345@somewhere.com SIP/2.0
...
Route: <sip:12345@10.1.1.5;lr>
my proxy will forward it to PSTN gateway and it will make outbound call.
Is this true? Please enlighten me on this.
Thank you,
Michael
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