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

[Olle E. Johansson]

6 aug 2009 kl. 15.24 skrev Iñaki Baz Castillo:

> 2009/8/6 Jan Janak <jan at ryngle.com>:
>> On Thu, Aug 6, 2009 at 2:42 PM, Iñaki Baz Castillo<ibc at aliax.net>  
>> wrote:
>>> 2009/8/6 Klaus Darilion <klaus.mailinglists at pernau.at>:
>>>>> At the same time, we could implement support for other URI's,  
>>>>> like XMPP
>>>>> since we have an xmpp gateway.
>>>>
>>>> Yes, should be generic as RFC 3261 which allows all kind of URIs
>>>
>>> Well, I can't agree. A SIP proxy shouldn't implement a HTTP URI in a
>>> request, or a mailto URI, even if RFC 3261 says "any URI".
>>
>> Why not?
>>
>>> AFAIK the only URI's to implement wouuld be:
>>> - SIP
>>> - SIPS
>>> - TEL
>>> - URN
>>
>> Why URN yes and HTTP not?
>
>
> 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).
Why should it be translated???

>
> But if SR impements HTTP perfect, then I'll configure a SR as HTTP
> proxy and load balancer XDD
>
No, but there's a lot of stuff now being implemented in HTTP requests  
in regards to SIP conferencing and SIMPLE.

/O