Channing the FROM, is it a good idea?

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Channing the FROM, is it a good idea?

thomcr
Hello List,

This question I think it's more related to SIP in general, or OpenSIPS - Asterisk implementation that just opensips. However I highly value the feedback and expertise of this list.  I have the following scenario.

Customers -> OpenSIPS -> Asterisk Farm -> PSTN

I use my OpenSIPS for multiple reasons, authentication, billing, authentication.

However the Asterisk servers see the source IP for all SIP traffic as the Proxy and not the customer IP. Which for the most part is ok. However limit the flexibility to the configuration, as I have to apply the same settings to all customer "hidden" by the load balancer.

I could overwrite the FROM, to force asterisk to match a profile based on the customer id, taken from an AVP. However Im worried about a few implications.

Some of my customer deliver me Caller id on the FROM, Horrible I know, so I copy it over to the PAI header and then manually parse the header in asterisk.

Would overwriting the FROM cause any problems? Is it ok acording to SIP RFC point of view? Would a SIP client freak out seen the FROM has changed on the Sequential replies for the dialog.

If changing the FROM is not a viable solution, what other alternatives do I have? Has anyone gone through this problem before and come up with some clever solution?

And this is not just Asterisk related, it could theoretically be any gateway. If there's a SIP proxy hiding the original Source IP/Port, which is the most commonly used configuration to match SIP trunk profiles in almost any equipment, then what can you use? What If I want to forbid a codec for a particular customer. 

I would be surprised if this issue has not come across an RFC or an implementation, but I have not read all of them, not quite ;)

Thank you for your feedback in advanced. 

--
Robert

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Re: Channing the FROM, is it a good idea?

Adrian Georgescu
Append a header in OpenSIPS with the source IP and check it in Asterisk.

Adrian
 
On Sep 11, 2011, at 4:30 AM, Robert Thomas wrote:

> Hello List,
>
> This question I think it's more related to SIP in general, or OpenSIPS - Asterisk implementation that just opensips. However I highly value the feedback and expertise of this list.  I have the following scenario.
>
> Customers -> OpenSIPS -> Asterisk Farm -> PSTN
>
> I use my OpenSIPS for multiple reasons, authentication, billing, authentication.
>
> However the Asterisk servers see the source IP for all SIP traffic as the Proxy and not the customer IP. Which for the most part is ok. However limit the flexibility to the configuration, as I have to apply the same settings to all customer "hidden" by the load balancer.
>
> I could overwrite the FROM, to force asterisk to match a profile based on the customer id, taken from an AVP. However Im worried about a few implications.
>
> Some of my customer deliver me Caller id on the FROM, Horrible I know, so I copy it over to the PAI header and then manually parse the header in asterisk.
>
> Would overwriting the FROM cause any problems? Is it ok acording to SIP RFC point of view? Would a SIP client freak out seen the FROM has changed on the Sequential replies for the dialog.
>
> If changing the FROM is not a viable solution, what other alternatives do I have? Has anyone gone through this problem before and come up with some clever solution?
>
> And this is not just Asterisk related, it could theoretically be any gateway. If there's a SIP proxy hiding the original Source IP/Port, which is the most commonly used configuration to match SIP trunk profiles in almost any equipment, then what can you use? What If I want to forbid a codec for a particular customer.
>
> I would be surprised if this issue has not come across an RFC or an implementation, but I have not read all of them, not quite ;)
>
> Thank you for your feedback in advanced.
>
> --
> Robert
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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