9 Sep
2006
9 Sep
'06
1:14 p.m.
Greger:
Thanks for addressing this. I've had a few questions about From
rewriting over the years. I'd like to add one more point. RFC3323 does
permit the display name and uri in the From header to be rewritten for
privacy concerns providing the tag is maintained. So rewriting the From
header to somthing like the following is acceptable for applications
such as caller ID blocking.
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=873345996
That being said I've moved away from this technique for caller ID
blocking because it presents problems with call detail record
generation. I generate CDRs in my gateways. With this form of caller ID
blocking the CDR record shows the called number but the calling number
is the word "anonymous". To get around this I now use Remote-Party-ID
headers with a display name of "anonymous" but a uri which preserves the
original calling number.
We are looking at using P-Asserted-Identity next to remain current
with the standards but we aren't there yet.
-Steve
Greger V. Teigre wrote:
Ok, I've written this before, but it's
important and worth another try.
First of all some clarifications:
- From and To have nothing to do with routing
- The requirement to not touch From and To has nothing to do with SER
implementation
- The tags in From and To together with Call-Id form a dialog and is
important for matching SIP messages in a transaction (ex. an INVITE -
OK - ACK sequence) and a dialog (ex. later reINVITEs, BYE etc)
http://www.tech-invite.com/Ti-sip-abnf-hf.html#from
Thus, in order to be RFC3261 compliant, you need to:
1. NOT touch the tags in From and To
2. NOT touch the remainder of From and To
If you break #1, you mess up transaction and dialog matching and
pretty much everything breaks. If you break #2, you violate the
requirement to be backwards compatible.
That being said, if ALL your UAs and gateways support and use RFC3261,
changing From and To names/uris (while leaving tags) will probably
work, but as all subsequent SIP related IETF documents and
implementation try to be RFC3261 compliant, it is assumed that From
and To are only changed two places:
a. In the UAs
b. In a B2BUA, i.e. a server that terminates a dialog with UA1 and
creates another one with UA2 and thus is a go-between
New SEMS can be used for b.
So, regardless of using uac module or subst for replacing From/To, you
will be MUCH better off if you follow the intentions of the RFCs, i.e.
change the From/To in the UAs through the provisioning systems you
use. Having lots of non-compliant SER installations is a recipe for
trouble in the future.
Note that there is an exception to the anonymization of a call. I
tried to find the reference, but I couldn't find it. Does somebody sit
on the reference?
I hope that was clarifying.
g-)
Ricardo Carvalho wrote:
It isn't necessary to restore those
from-header fields to maintain
the call leg of transactions taking place. As far as I know, by the
tests I made, Ser routes SIP messages based on Call-Id and not as
well From and To tags of messages.
Although this is only true if all UAs that you use are
RFC3261compliant. If some phones that you use implement the old
protocol, uses the full content of From/To for the same purposes, you
may get some problems... It's risky manipulate From/To tags...
Regards,
Ricardo.
G.Jacobsen wrote:
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Ricardo,
Ok, but how do you restore the from-header when you send answers of the
downstream party to the upstream party again ?
Cheers
Gerry
----- Original Message ----- From: "Ricardo Carvalho"
<rcarvalho(a)iric.up.pt>
To: "Alex Fler" <alexfler(a)yahoo.com>
Cc: <serusers(a)lists.iptel.org>
Sent: Friday, September 08, 2006 5:15 PM
Subject: Re: [Serusers] Re: rewrite the FROM part of SIP INVITE
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I've used subst() function for substituting
From and To URIs and calls
succeed well! I think that this indicates that Ser uses in fact
only the
Call-Id to keep track of calls and not as well From and To tags of
messages. Although I this may end up biting me later...
You can do that for example with the following syntax:
subst('/^From:(.*)sip:.*@your_domain(.*)/From:\1sip:what_ever_number@what_ev
er_domain\2/');
Greetings,
Ricardo.
Alex Fler wrote:
> Hello guys,
>
> Sorry for a stupid question, but how do you actually rewrite the FROM
> part of SIP INVITE ?
>
> Do you use avp ? Could someone give me an example ?
>
>
>
> Thanks to all
>
>
>
> Alex Fler
>
>
>
> ------------------------------------------------------------------------
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>
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>
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