Naturally, either way is stupid.<br><br>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.<br>
<br>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.<br><br>This is, of course, the first step into a lengthy 12 hour discussion on how stupid SIP and url/text based<br>
protocols are.<br><br>I dare someone to crank up the pcap on a box doing SIP presence for 20 phones and "read"<br>the 1200 byte messages with all kinds of hyeroglyphic url syntax and embedded xml payloads and write<br>
up a paper on how much "sense" it makes to have it be "readable".<br><br>PS<br><br>supposedly sofia can support sctp,<br>someone should try it.<br><br><br><br><div class="gmail_quote">On Mon, Dec 1, 2008 at 9:43 PM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kkielhofner@star2star.com">kkielhofner@star2star.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">On 12/1/08, Thomas Troy <<a href="mailto:ttroy50@gmail.com">ttroy50@gmail.com</a>> wrote:<br>
..snip..<br>
><br>
> Out of interest do you have any links to anywhere this is discussed in terms<br>
> of general sip implementations?<br>
><br>
<br>
Uh oh, here we go again...<br>
<br>
<a href="http://www.iana.org/assignments/sip-parameters" target="_blank">http://www.iana.org/assignments/sip-parameters</a><br>
<a href="http://tools.ietf.org/html/rfc3969" target="_blank">http://tools.ietf.org/html/rfc3969</a><br>
<br>
<a href="https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-August/010047.html" target="_blank">https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-August/010047.html</a><br>
<br>
Implementation wise, most devices tend to use transport=tls:<br>
<br>
SIPFoundry - From what I've seen<br>
Snom<br>
SERs<br>
Asterisk (If you are using TLS)<br>
Cisco - I *believe* you can use either a SIPS URI or the transport=tls<br>
parameter for various SIP targets<br>
<br>
As the RFC (basically) states (RFC3261, section 12.1.x),<br>
transport=tls was deprecated in RFC 3261 because you should also be<br>
able to do TLS over SCTP (RFC3436), which makes transport=tls a bit<br>
ambiguous. sips:user@domain;transport=tcp or<br>
sips:user@domain;transport=sctp is a bit more flexible.<br>
<br>
I don't know if I've ever seen anything default to SIPS URIs. I<br>
also don't think I've ever specifically tried using them. However, my<br>
experience with TLS is admittedly somewhat limited so this shouldn't<br>
be taken as gospel. As you can see from the discussions on<br>
sip-implementors, this gets interesting when different devices are<br>
traversing a proxy using different URI schemes...<br>
<br>
However, I suspect this won't become an issue until most SIP<br>
implementations support SCTP. That should be exciting! ;)<br>
<br>
<br>
--<br>
Kristian Kielhofner<br>
<a href="http://blog.krisk.org" target="_blank">http://blog.krisk.org</a><br>
<a href="http://www.submityoursip.com" target="_blank">http://www.submityoursip.com</a><br>
<a href="http://www.astlinux.org" target="_blank">http://www.astlinux.org</a><br>
<a href="http://www.star2star.com" target="_blank">http://www.star2star.com</a><br>
<br>
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