From the wiki: "There IS a wrinkle to this special case: If you have a URL of the form <sip:user@example.com:5055> the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host example.com (i" This is an error in the RFC that we are trying to fix in the IPV6 wg of the SIP Forum. Software should NOT find A *OR* AAAA. It should look up both and then try them according to Happy Eyeballs - in parallell with a small advantage for IPv6. If we do serial forking here, we have in worst case a 32 second timeout in SIP over UDP, which is not acceptable. Every customer should require fixes and every developer should try to fix this... Some slides on these issues: * http://www.slideshare.net/oej/sipforum-sip-ipv6-discussion-slides * http://www.slideshare.net/oej/sip-ipv6-time-for-action I hope to see FreeSwitch at SIPit 30 in February so we can test this and other issues! /O 9 jan 2013 kl. 00:21 skrev Michael Collins <msc@freeswitch.org>:
I couldn't stand it, so I added Lawrence's info to this page: http://wiki.freeswitch.org/wiki/DNS#Explanation_of_DNS.2FSRV.2FNAPTR
Cheers, MC
On Tue, Jan 8, 2013 at 2:56 PM, Michael Collins <msc@freeswitch.org> wrote: Lawrence,
Thanks for this explanation. It was very well written. I'm looking for a place to codify this on the wiki so that it gets preserved... :)
-MC
On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy <lconroy@insensate.co.uk> wrote: Hi there, at the risk of butting in on someone else's party ... Nope; your interpretations is NOT best practice. I have some sympathy, as the term domain is overloaded within fS.
A sip address consists of a userpart and a domain part -- e.g., <sip:user@sipdomain> The sip domain is similar to an email domain -- e.g., <mailto:user@maildomain> With email, you need to do a lookup of the MX record in DNS to find the FQDN of the machine that handles mail for the domain. With SIP (see RFC 3263), you do a lookup on the SRV record (at _sip._udp.<sipdomain>) to find the machine that handles SIP registrations/incalls for the domain. That also gives you the port on which that machine is listening. (Yup, you can also have a NAPTR record in the domain to tell you where the SRV record is, but many folks don't bother -- for Best Practice, you should, but ...)
There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) support that means most SIP clients will look for the SRV and, if it can't be found (or there's an IP address rather than a DNS -style domain, in which case the SIP client won't bother hunting the SRV), the client will guess that the domain has a machine (i.e. it will look for an A or AAAA record), and also guess it's listening on 5060 (the default port). Email is the same (mail to fred@example.com, and strictly the sender will do a check for a MX and then look for an A record for example.com, and try there).
However, relying on that default "get out" clause is definitely NOT what you should do for BCP. Using the hostname as the sip domain is a kludge -- the FQDN with A record usually works, but it's not what you want to do.
SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at _sip._udp.<domain>, and you're done. No need for an A record at that domain at all.
(RFC 3263 is not too hard to read, for a change -- it's certainly shorter than RFC 3261, and it even has an ASCII art diagram :).
all the best, Lawrence
On 8 Jan 2013, at 21:05, Steven Schoch wrote:
On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < fs-list@communicatefreely.net> wrote:
Hi Steven,
I would recommend using a proper domain name as much as possible. For one, it looks nicer! A SIP URI is supposed to be user@domain like an e-mail address is, and I hope that one day URI dialing will be common place, so we might as well do it right the first time.
What you're saying is that "domain" should really be a fully-qualified host name that points via DNS to the actual host on which FreeSwitch is running. That is, the domain should be "pbx.example.com" instead of just " example.com", as the last example would most likely point to a web server, not the SIP server. Do I have that right?
Next, in the configuration for Polycom phones (for example), there are 2 fields that both have the userid. In the example in http://wiki.freeswitch.org/wiki/Polycom_configuration it has:
reg.1.auth.userId="1000"
and
reg.1.address="1000@fs.domain.local"
How is the "address" value used? Is that sent in the SIP registration message? If that's the case, what does Freeswitch do with it?
-- Steve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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