[sr-dev] About SIP and TEL uri in SR

[Jan Janak]

On Thu, Aug 6, 2009 at 3:47 PM, Iñaki Baz Castillo<ibc at aliax.net> wrote:
> 2009/8/6 Martin Hoffmann <martin.hoffmann at telio.ch>:
>> Iñaki Baz Castillo wrote:
>>>
>>> According to some exotic RFC, a proxy should handle a URN URI and
>>> translate it into a SIP URI (or route the request to a predefined
>>> proxy which handles it). But no specification defines how a HTTP URI
>>> should be translated into a SIP URI (or other kind of URI).
>>
>> Because it isn't specified, it can't be done? One could probably think
>> of some scenarios where a MESSAGE request with a text body and a mailto
>> URI does make sense. Of course you need some pre-configured logic to
>> make the proxy understand what to do with it. If it knows where to send
>> the request to, it can happily do that. No need to have a SIP URI for
>> that purpose.
>
> Ok, but for sure that's not the purpose of a SIP proxy ;)

Just to make things a bit more clear, I was asking why not because I
assumed that you were referring to some existing internet draft or RFC
which says that HTTTP (and possibly other URI types) should not be
used in SIP messages.

Possibly because they are considered harmful, or something like that
(IETF does issue such RFCs from time to time). Not being able to
recall any such document, I thought I missed it.

If that's not the case then I just mentally add "in my opinion" to the
original text to make is sound less definitive.

I was not trying to argue that HTTP (or any other URI types) should be
explicitly supported by a SIP proxy.

  Jan.