[asterisk-dev] Peer matching and SRV records

Fri Apr 19 09:56:44 CDT 2013

19 apr 2013 kl. 16:41 skrev "Olle E. Johansson" <oej at edvina.net>:

> 
> 19 apr 2013 kl. 16:30 skrev Eric Wieling <EWieling at nyigc.com>:
> 
>> I experienced similar issues.   Solved it in "userspace" rather than in the code.    For inbound we switched to username auth, though that is not an option for most use cases.   For outbound I wrote an AEL function using SRVLOOKUP to handle failover and priorities, based on the sample in the relevant RFC.
>> 
>> Are you convinced handling outbound SRV stuff in Asterisk is the right way, as opposed to handling it in the dialplan?
> 
> Yes. Very few people can build it properly in the dialplan and using the dialplan we can't benefit from the settings in the
> sip.conf peers - which is quite often needed, like fromdomain, fromuser, codecs, authentication and much more.
> 
Forgot one benefit:

With proper SRV support, the INVITE can go to one server and the BYE  (or any other in-dialog message) to another, 
which is critical in large SIP proxy installations. You can not do that in the dial plan.

/O
> 
> I will in no way cancel the functions that we have in the dialplan , but I don't think that's the right way to go for most of the Asterisk user base.
> 
> Normal users wants to configure a SIP service they pay for and expect Asterisk to handle the failover properly. We should deliver that.
> 
> (And besides - that's what my customer pays me for :-) )
> /O
> 
> 
>> 
>> ---
>> Documentation for Polycom phones can be found at http://help.nyigc.net/
>> 
>> ________________________________
>> From: asterisk-dev-bounces at lists.digium.com [asterisk-dev-bounces at lists.digium.com] On Behalf Of Olle E. Johansson [oej at edvina.net]
>> Sent: Friday, April 19, 2013 10:26 AM
>> To: Asterisk Developers Mailing List
>> Cc: Olle E. Johansson
>> Subject: Re: [asterisk-dev] Peer matching and SRV records
>> 
>> 
>> 19 apr 2013 kl. 16:17 skrev Jaco Kroon <jaco at uls.co.za<mailto:jaco at uls.co.za>>:
>> 
>> Hi Olle,
>> 
>> I can confirm your observations.  For this reason I've updated (or initiated the process) to periodically resolve the SRV records out-of-band and generate an updated configuration file that contains all of the hosts.  I must further point out that (last I checked) it gets even worse, if you have an A record with multiple IPs listed, asterisk will only use one of those as well.
>> 
>> My scenarios only have one IP in the A RRs but I do have multiple SRV RRs for all of _sip._udp, _sip._tcp and _iax._udp.
>> Ok, did not think of that. Hmm.
>> 
>> 
>> To add further insult to injury, an *outbound* Dial() also won't use secondary SRV RRs, it'll only go to the first "picked" host, and only use that, so even if timers expires and we're unable to contact the remote side ... the call WILL fail.  One way to do this would be to construct a list of contact points, and for SIP at least only use the T1 timer to cycle through them with INVITEs, first one to respond ... grabs it, if they all fail, send to all of them at the same time with the T2 timer, first one to respond grabs it, any other responses simply gets CANCELed.  I can see a great many number of holes in this strategy, but it may be a starting point for someone else to start thinking from.
>> 
>> That's the stuff I'm targeting in SIP now. We should fail over as long as we have SRV priorities left.
>> 
>> This applies to both SIP and IAX/2 (and probably other protocols that I don't even know about).
>> SIP is my focus. ;-)
>> 
>> Currently my suggestion would be to deal with the outbound situation in Dialplan(), and to generate multiple peer configs in the config files, covering each host individually.  Obviously this implies that you have to track DNS changes external to asterisk.  My setup also utilizes the inbound ACLs for dealing with IAX/2 authorization (so all peers will send IE username=foo, and [foo] will deny=all, permit=${ips_from_srv_and_a_rrs}).  Still need to generate multiple peers for outbound cases though.
>> 
>> Please check the README for the new branch where I have jotted down some ideas. We have the dialplan support for resolving SRV records, I want to make it automatic in the SIp channel for both DIAL() a peer with SRV records and getting calls from them.
>> 
>> http://svnview.digium.com/svn/asterisk/team/oej/pgtips-srv-and-outbound-stuff-1.8/README.pgtips-srv-records
>> 
>> No code yet, just moving around preparing stuff.
>> 
>> 
>> If you'd like to fix this in the code - something which should probably done, and would be a better solution than my hack, but which I suspect is going to be rather complex - I'd be more than willing to help test for you.
>> Thank you for feedback. And thanks for the feedback on RTCP - will go through it soon.
>> 
>> The funding for this project is from Inteno Broadband Solutions. Proper SRV record support in Asterisk is something I wanted for years.
>> 
>> /O
>> 
>> Kind Regards,
>> Jaco Kroon
>> 
>> On 19/04/2013 11:48, Olle E. Johansson wrote:
>> 
>> Friends,
>> 
>> Looking into the SRV record support of Asterisk I believe there's an issue with peer matching here.
>> 
>> If a service, like "edvina.org<http://edvina.org>", have multiple SRV records that points to multiple hosts IPs, maybe even dual stack, then a request FROM that service
>> may come from any of these IPs.
>> 
>> Let's assume the configuration looks like this:
>> 
>> register=marko:okram at edvinaservice/callback
>> 
>> [edvinaservice]
>> type=peer
>> host=edvina.org<http://edvina.org>
>> 
>> 
>> In this case the peer will pick one address from the SRV records and use for matching. If another server IP is used on the
>> server side, matching will fail.
>> 
>> I would like to be able to add all available IP addresses and ports for matching. Will that work with the ao2object list or will it
>> mess up the list to have many hash entries for the same object?
>> 
>> The way I would like to do this is to set up an ACL entry in the peer for the SRV record so we have a list to go through
>> and perform the matching on. If that list is empty, we will match as before.
>> 
>> Thoughts?
>> /O
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