On 5/21/13 11:00 AM, Olle E. Johansson wrote:
21 maj 2013 kl. 10:52 skrev Alex Balashov <abalas...@evaristesys.com>:
The problem one always runs into when dealing with phones, softphones, ATAs,
and other end-user clients is getting them to trust incoming calls from the
secondary registrar in the event of a failure of the primary one. Usually,
they expect calls to come from the registrar that they registered to, and few
of them implement SRV correctly enough to solve this problem.
Well, that's why we are forced to use IP failover. Sad but true.
This is not an issue with PBXs, where one can generally build two trunks to two
different hosts.
Asterisk has been completely broken in regards of SRV support. It doesn't fail
over, it doesn't handle IPv6 and Ipv4 in SRV priorities correctly, and it
doesn't accept incoming calls from all hosts in the SRV record set.
I'm trying to fix that in my pgtips branch.
I don't know the state of FreeSwitch in regards of SRV support, nor do I know
what's planned for the new SIP stack in Asterisk.
In Kamailio - is there any way that we can check if a server that contacts us
is part of a SRV record set?
If I get a call from a server using the domain @evaristesys.com - can I check
if that server actually is defined as authorative for that domain by checking
SRV records - and without a TLS client cert?
A dns-ops like module is somewhere in a dusty to-do list, at this moment
a solution would be to use lua or other embedded language.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
