[Asterisk-Users] Clustering

Sun Mar 12 22:24:58 MST 2006

Im sure the ServerIron can do that, but I cant believe its SIP aware and
actaully tears apart the SIP packet and then re-assembles it with the right
info. (In theory what a SBC is suppose to do). I just checked Foundrys
website again, and I see no mention of SIP, I just see the ServerIron doing
SSL offloading, nothing like packet rewriting though. So I think we are back
to SER or a SBC from someone...

Thanks!
Ron

On 3/12/06, Gabriel Afana <asterisk at gafana.com> wrote:
>
> On a side note, the ServerIron can do Reverse-Nat where it will rewrite
> the source IP to its Virtual IP and when requests return back, it routes it
> back to the same server/port.  It can actually do a great deal of things,
> this is why I am sure there has to be a way to get this done.
>
> - Gabe
>
>
> ----- Original Message -----
> *From:* Wai Wu <wwu at Calltrol.com>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at lists.digium.com>
> *Sent:* Sunday, March 12, 2006 5:42 PM
> *Subject:* RE: [Asterisk-Users] Clustering
>
> Ron,
>
> Think the discussion has drifted a bit. Looking back at your original
> post. What you wanted was a simple load lalancer to distribute the calls
> from registered sip phones across multiple servers. I think you can
> accomblish this with a script in the entry extension (on the master server)
> that pulls for CPU utilization of the other servers and send the call to the
> one that's least utilized. As for RTP packets. I thanks the 'canrevite'
> scheme in * can handle it automatically, i.e. RTP packets will bypass the
> master server and directly to the call processor server.
>
> -----Original Message-----
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com]*On Behalf Of *Ron McCarthy
> *Sent:* Sunday, March 12, 2006 5:07 PM
> *To:* Gabriel Afana; Asterisk Users Mailing List - Non-Commercial
> Discussion
> *Subject:* Re: [Asterisk-Users] Clustering
>
> Hi Gabe,
> Well im guessing your ServerIron wont work because its not near smart
> enough to know how SIP works, let alone, 99.9% of load balancers I have
> seen use private IP's on server side, the and Load Balncer then has the
> public IP's assigned to them. Right there, this creates a problem in itself.
> But im assiumg the Foundry isnt smart enough to keep track of multiple
> phones from the same IP, and all the RTP sessions associated with it, since
> like you said, several hundred port numbers are being used. The Juniper box
> seems to rewrite the actual SIP header on the outbound transversal to the
> Internet, this solving the NAT return path problem, and then it keeps track
> in a state table as to what ports go to what server, etc, etc. But I think
> there is no way this could "failvoer" in the middle of the car, since it
> would somehow have to change the RTP stream to another port, but also the
> phone would have to get to get registered on that server as well, which its
> not, which is why Douglas is using SER to have it register on several
> different machines, so when the failover occurs the phone is registered and
> the RTP stream just needs to pick up. Im trying to see exactly how he is
> doing this, since this is the exact thing I need, and then Ill just run OSPF
> on my core router (not sure if that will work yet).
>
> I woudl perfer to do this all in hardware vs software since a
> Cisco/Juniper box is musch less prone to failure then a server with
> software, but I guess more research will tell what ill be using in the end
> :)
>
> Once I get this going, I want to post a entire howto on the wiki.
>
> Thanks!
> Ron
>
> On 3/12/06, Gabriel Afana <asterisk at gafana.com> wrote:
> >
> >  Hi Ron,
> >     If the SBC would have served mearly as a load balancer...I already
> > have one and it didn't work too well.  I have a Foundry ServerIron XL load
> > balancer and I've tried using it with Asterisk.  It has had positive and
> > negative results.
> >
> > Positive:  It *would* load balance between asterisk servers for whatever
> > port I set (I was using 5060 for SIP).  However, I didn't mess with the RTP
> > because its got so many ports (and you can't add ranges for virtual server
> > ports, you have to enter exact ports - at least I think) and because I have
> > no idea how that would work if SIP signaling goes to one server and RTP goes
> > to another???  (probably not!)  I would create a virtual IP on the load
> > balancers and have all the phones register to this IP.  When checking status
> > of the ports on each server, it showed 5060 for all servers was unused (0
> > current connections).  When I would make a call, it would show the 5060 port
> > on one of the * servers in use (1 current connections) and it worked
> > fine....this is where the problem started.
> >
> > Negative:  When I would hang up the phone, it would still show 1 current
> > connection to that server's 5060 port.  Every call I would make from then on
> > would *still* go to that same server.  It seems the ports are "sticky" or
> > set with a keepalive.  Of course I can define these options on the
> > ServerIron, but even with "sticky" disabled and keepalive disabled, the port
> > would appear active (like keepalive was enabled) and every call would go to
> > the same server (like "sticky" was enabled).  Even if I would shutdown
> > asterisk on that server, it would still show an active user on that port and
> > when I would make the call, the call would not go through.  The LB was not
> > failing the port.  I think maybe if I keep playing with it...?   Any
> > suggestions?
> >
> > If I can get my ServerIron working, I will do a complete write up on
> > it...but it works only partially.
> >
> > This is why I was so interested in the Juniver SBC....if it would be
> > able to act a proxy, do all the load balancing and instantly failover if a
> > server fails; basically a VoIP Load Balancer.  But I guess thats not what it
> > does.  Does a VoIP load balancer hardware exist or is the only solution
> > right now software proxies like SER?
> >
> > - Gabe
> >
> >
> >
> >  ----- Original Message -----
> > *From:* Ron McCarthy <ronmccar at gmail.com>
> >  *To:* Gabriel Afana <asterisk at gafana.com> ; Asterisk Users Mailing List
> > -Non-Commercial Discussion <asterisk-users at lists.digium.com>
> > *Sent:* Sunday, March 12, 2006 1:16 PM
> > *Subject:* Re: [Asterisk-Users] Clustering
> >
> > Hi Gabe,
> > Well I was going to use the SBC to have all phone point to the SBC, and
> > then the SBC takes care of what servers it needs to register with, and then
> > keep a state of what server the RTP stream and the phone need to connect to.
> > Basically like a load balancer would. This is what I understood from
> > Juniper's site. Have you seen anything on this?
> >
> > Thanks!
> > Ron
> >
> > On 3/11/06, Gabriel Afana <asterisk at gafana.com> wrote:
> > >
> > >  Hi Ron,
> > >     I've been following your thread.  I noticed you mentioned about a
> > > Juniper Session Border Controller.  I checked online and read about it, but
> > > was unsure exactly how it could intergrate with Asterisk.  How would you
> > > have planned to use that device?  I am interested because one of my upstream
> > > providers mentioned I should be using an SBC.
> > >
> > > - Gabe
> > >
> > >
> > >  ----- Original Message -----
> > > *From:* Ron McCarthy <ronmccar at gmail.com>
> > >  *To:* Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at lists.digium.com>
> > >  *Sent:* Friday, March 10, 2006 11:22 AM
> > > *Subject:* [Asterisk-Users] Clustering
> > >
> > > Hello All,
> > >
> > > Ive been doing more and more research on trying to setup a
> > > cluster/load balancer for Asterisk. All the Asterisk boxes would be using a
> > > config that is the same between them all (via a DB), but we want one
> > > location to point the phones to, and from there that machine/device will
> > > send it to a Asterisk server so the call can be processed. I know you cant
> > > balance the whole call, ie: once the call is started the RTP stream has to
> > > go to the same server, but a new call could go to a different server if
> > > perhaps the 1st server was unreachable.
> > >
> > > Has anyone tried this, or got this to work? Ive been looking at using
> > > a Juniper Session Border Controller, but not sure if thats gonna do the
> > > trick, and then we also have SER..
> > >
> > > Any comments would be great!
> > >
> > > Thanks
> > > Ron
> > >
> > > ------------------------------
> > >
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> >
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