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:
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
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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