[Serusers] Re: rewrite the FROM part of SIP INVITE

Steve Blair blairs at isc.upenn.edu
Sat Sep 9 12:14:46 CEST 2006


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 at 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:
>>
>>> 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 at iric.up.pt>
>>> To: "Alex Fler" <alexfler at yahoo.com>
>>> Cc: <serusers at 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 at 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
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------ 
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Serusers mailing list
>>>>> Serusers at lists.iptel.org
>>>>> http://lists.iptel.org/mailman/listinfo/serusers
>>>>>
>>>>>       
>>>>
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>>>>     
>>>
>>>
>>>
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>>
>>
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