On Apr 23, 2014, at 8:42 AM, Antonio Querubin tony@lavanauts.org wrote:
(Please trim inclusions from previous messages) _______________________________________________
On Wed, 23 Apr 2014, Danny Messano wrote:
That's ENUM, not DNS SRV, and again, not necessary. I know of very few software SIP clients that don't allow SIP URI dialing. No need for numerics whatsoever. I can dial alphanumeric user@host from any number of clients.
No this has nothing to do with ENUM. In general, one cannot absolutely assume name@domain is reachable at a SIP server running on a host named 'domain'. That's inflexible and is like assuming only host 'domain' will accept email for name@domain. There may not even be an actual host named 'domain' or if there is, it may not be running SIP. Ignoring SRV RRs also means your SIP client/proxy or SBC wouldn't be taking advantage of the resiliency provided by alternate target servers.
I think there's some wires crossed up here.
You brought up the notion of dialing somehorriblestringofnumbers@ampr.org to reach a user, which is entirely where ENUM comes in. We don't need any of that, for the aforementioned reasons.
Now back to the real need for DNS SRV records. Yes, I addressed much earlier that in SOME instances there may be a need to locate my SIP resource somewhere other than my call.ampr.org hostname. But that isn't the same as "we need DNS SRV to find each other or else", and won't be the case 99.99% of the time.
Now as far as the redundancy and failover aspects... We took a pretty big jump here from "I want to set up Asterisk and call my ham buddies over 44net" to "You need DNS SRV if you're setting up redundant SBCs". What is the goal here? Is everyone going to set up multiple SBCs and we're going to built out a 44net AT&T (I dont see anyone doing the work) or do we need a simple schema in place for all of us to happily dial peer-to-peer SIP and really play with this stuff?
Sometimes I feel like these discussions are networking pissing contests and nothing to do with actually implementing something NOW. If that's the case, RFC 3261 explains everything i've discussed.
Many of us have been doing SIP URI dialing using softphone and hardware endpoints for years with just a little dialplan work in Asterisk or an appropriate softphone. We could start that today on 44net or continue to engineer this ad nauseum.
Danny Messano
Antonio Querubin e-mail: tony@lavanauts.org xmpp: antonioquerubin@gmail.com _________________________________________ 44Net mailing list 44Net@hamradio.ucsd.edu http://hamradio.ucsd.edu/mailman/listinfo/44net