[Freeswitch-users] TLS receiving calls

[Anthony Minessale]

Naturally, either way is stupid.

The whole idea of putting the transport in a uri param is equally stupid to
using 2 different protocol names but since SIP is the descendant of http it
they decided to stick with the stupidity of http/https and have sip/sips
which is almost as if it was designed to break all software trying to keep
up with url syntax.

If they are going to insist on using text params you'd think something like
transport=foo;security=tls would be even *more* flexable in case alternate
methods to encrypt crop up.

This is, of course, the first step into a lengthy 12 hour discussion on how
stupid SIP and url/text based
protocols are.

I dare someone to crank up the pcap on a box doing SIP presence for 20
phones and "read"
the 1200 byte messages with all kinds of hyeroglyphic url syntax and
embedded xml payloads and write
up a paper on how much "sense" it makes to have it be "readable".

PS

supposedly sofia can support sctp,
someone should try it.



On Mon, Dec 1, 2008 at 9:43 PM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On 12/1/08, Thomas Troy <ttroy50 at gmail.com> wrote:
> ..snip..
> >
> > Out of interest do you have any links to anywhere this is discussed in
> terms
> > of general sip implementations?
> >
>
> Uh oh, here we go again...
>
> http://www.iana.org/assignments/sip-parameters
> http://tools.ietf.org/html/rfc3969
>
>
> https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-August/010047.html
>
> Implementation wise, most devices tend to use transport=tls:
>
> SIPFoundry - From what I've seen
> Snom
> SERs
> Asterisk (If you are using TLS)
> Cisco - I *believe* you can use either a SIPS URI or the transport=tls
> parameter for various SIP targets
>
>  As the RFC (basically) states (RFC3261, section 12.1.x),
> transport=tls was deprecated in RFC 3261 because you should also be
> able to do TLS over SCTP (RFC3436), which makes transport=tls a bit
> ambiguous. sips:user at domain;transport=tcp or
> sips:user at domain;transport=sctp is a bit more flexible.
>
>  I don't know if I've ever seen anything default to SIPS URIs.  I
> also don't think I've ever specifically tried using them.  However, my
> experience with TLS is admittedly somewhat limited so this shouldn't
> be taken as gospel.  As you can see from the discussions on
> sip-implementors, this gets interesting when different devices are
> traversing a proxy using different URI schemes...
>
>  However, I suspect this won't become an issue until most SIP
> implementations support SCTP.  That should be exciting! ;)
>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
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-- 
Anthony Minessale II

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