SV: SV: [Serusers] Handling 302 responses

Tue Sep 12 17:47:09 CEST 2006

Thank you for your answer.

You are probably right that I would get better answer without the "off the
wall" remark. Sorry if upset anyone.

Ok, now we are getting some where. The 302s I get back from the UAs contains
a diversion header with the callee info and the contact header contains the
destination for the "forwarding". So is it possible to extract the info from
these headers to use when appending the branch?

Kind regards
Roger

  _____  

Från: Greger V. Teigre [mailto:greger at teigre.com] 
Skickat: den 12 september 2006 08:51
Till: Roger Lewau
Kopia: serusers at iptel.org; jh at tutpro.com
Ämne: Re: SV: [Serusers] Handling 302 responses

Hi Roger,
I think you are complicating the picture by trying to join two quite
different worlds: SIP world and the PSTN world.  SIP was not intended to be
introduced into the PSTN world, it applied some basic "Internet-type"
principles to the problem of setting up media session over the Internet.
Even the new IMS specifications for 3G follow the intentions of SIP
(although there are some conflicts between "telco people" and "Internet
people".  

By no means are you alone, many UAs have been implemented with PSTN
replacement services in mind.  In PSTN, call forwarding is associated with
multiple call legs, each leg being charged to the A-number (calling party).
(Of course, tromboning can can be done to optimize the media path, but
signalling-wise the call legs are still present, and thus also in billing).
In SIP, a REFER is not the creation of a new call leg, it is simple a
message back to caller that callee can be reached elsewhere. Strictly
speaking, to simulate PSTN, but using SIP intentions, the SIP UA should
implement B2BUA and initiate a new INVITE to the forwarded address. The
reason is that SIP puts more responsibility on the endpoints than in PSTN
(where the endpoints are dumb). Of course, you don't want to do this.

    So, to your problem: You probably have incoming PSTN calls to your UA,
which you have decided should use 302 for "forwarding". The 302 is a reply,
not a failure, so you will receive it in your onreply route.  You can
probably then test for 302, change the ruri and append a branch. Depending
on your billing handling, you should add a Diversion header, CC-Diversion or
store in acc module using avpairs (Jiri noted that this is possible in a
recent thread, I have never done it) the party who redirected.

Maybe these call flows can help in understanding 302. It doesn't cover 302
from clients though.
http://www.cisco.com/univercd/cc/td/doc/product/voice/sipsols/biggulp/bgsipc
f.htm

I hope this makes sense to you.

BTW, I think you would received some sensibles answers earlier if you hadn't
used your "off the wall" statement...
g-)
Roger Lewau wrote: 

Hello Greger

I am surprised that you argue that 302s is not the standard call forward
scenario when this is how it is implemented in almost all sip equipment
(haven’t seen any device who does it differently). Customer will punch in
cfwd (like *72<number>) in their phone and expect the operator (me) to
forward the call to the destination at their expense. This is nothing new or
strange. 302s signal that an UA has temporarily been moved and can be
reached somewhere else, how is that not call forwarding?  Also, I would
never expect an UA to send back an email address in an IP telephony system,
even if it is theoretically possible. I would expect it to send back a SIP
address; everything else is not reachable through the protocol of SIP and
makes absolutely no sense.

Most Voip customers are not techies and they are not rfc knowledgeable and
could not care any less about how the forwarding is made. They would not be
upset that their 302 generate a cost, this is the expected behaviour for
this service.

Kind regards

Roger

  _____  

Från: serusers-bounces at lists.iptel.org
[mailto:serusers-bounces at lists.iptel.org] För Greger V. Teigre
Skickat: den 11 september 2006 11:59
Till: Roger Lewau
Kopia: serusers at iptel.org; jh at tutpro.com
Ämne: Re: [Serusers] Handling 302 responses

Hi Roger,
I think that was Juha's point: we don't. 
302 was created to enable a user agent to communicate back to the other user
agent that it can be reached somewhere else. Thus, your server should relay
the 302 and the receiving user agent should then decide what to do. Some UAs
immediate initiate a new call, while others (e.g.software agents) may pop up
a question to the user: "Callee is not available, but can be reached at
location" (which of course may well be an international PSTN call that can
be expensive).
Some UAs also have options that can be set: How to handle redirects

Server-centric forwarding can be better handled by user preferences and
loading av pairs.

That being said, I remember a thread a while ago with a discussion on how to
turn a 302 into a forwarding. I don't remember the outcome, but it is
probably possible, although not according to the RFCs.  You do have some
problems though, e.g. if the UA sends back an email uri etc.

And of course, as people tend to follow RFCs, you will probably get one
angry customer if he realizes that his 302 generates a cost. If you have
control over the UA and have decided to use 302 instead of the more
standardized call forward scenario, you  really are making problems for
yourself.
g-)

Roger Lewau wrote: 

Hello Juha and Andrey

302 "Moved temporarily" is definately about forwarding/redirecting calls.
This is how the vast majority of all IP phones and ATAs handle call
forwarding. It might not be the intended use of 302 according to RFCs, even
if I see nothing that says otherwise, but this is how it is used in end
devices today. This brings us back to my original question. How do you guys
handle 302 redirection so that costs are charged to the callee.

Kind regards

Roger 

-----Original Message-----
From: "Andrey Kouprianov"  <mailto:andrey.kouprianov at gmail.com>
<andrey.kouprianov at gmail.com>
To: serusers at iptel.org
Date: Mon, 11 Sep 2006 12:51:09 +0700
Subject: Re: [Serusers] Handling 302 responses

You can also use 302 responses to gather some information about the
remote party. Contacts returned in the response are not necessarily
the SIP URI's. I've tried using mail addresses, SIP tel: URI's and
HTTP URLs too.

So, if the remote party is Busy at the moment, but has other ways to
let u contact them, 302 is one of the answers to this.

On 9/11/06, Juha Heinanen  <mailto:jh at tutpro.com> <jh at tutpro.com> wrote:
> Roger Lewau writes:
>
>  > In my mind that statement is completely off the wall, it is not the
>  > requesting client that should be responsible for establishing the
forwarded
>  > call, it never is in the rest of the telecom industry so why should it
be
>  > the case for SIP?
>
> 302 is not about "forwarded call".  it just tells the caller that the
> callee is at some other uri, which the caller may or may not wish to
> contact.  in many pstn networks, you can hear an announcement that the
> number you tried is not in use and you should try another number
> instead.
>
> if callee wants to "forward" calls, he has other means for that purpose,
> for example, his phone can forward the invite to some other uri or he
> may configure his proxy to do so.
>
> -- juha
> _______________________________________________
> Serusers mailing list
> Serusers at lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
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