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<DIV><FONT face=Arial size=2>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.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>- Gabe</FONT></DIV>
<DIV> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=wwu@Calltrol.com href="mailto:wwu@Calltrol.com">Wai Wu</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">Asterisk Users Mailing List -
Non-Commercial Discussion</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Sunday, March 12, 2006 5:42
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [Asterisk-Users]
Clustering</DIV>
<DIV><BR></DIV>
<DIV><SPAN class=401112301-13032006><FONT face=Arial color=#0000ff
size=2>Ron,</FONT></SPAN></DIV>
<DIV><SPAN class=401112301-13032006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=401112301-13032006><FONT face=Arial color=#0000ff
size=2>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.</FONT></SPAN></DIV>
<BLOCKQUOTE>
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]<B>On Behalf Of </B>Ron
McCarthy<BR><B>Sent:</B> Sunday, March 12, 2006 5:07 PM<BR><B>To:</B>
Gabriel Afana; Asterisk Users Mailing List - Non-Commercial
Discussion<BR><B>Subject:</B> Re: [Asterisk-Users]
Clustering<BR><BR></FONT></DIV>Hi Gabe,<BR>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).<BR><BR>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 :) <BR><BR>Once I get this going, I want to post a entire howto
on the wiki.<BR><BR>Thanks!<BR>Ron<BR><BR>
<DIV><SPAN class=gmail_quote>On 3/12/06, <B class=gmail_sendername>Gabriel
Afana</B> <<A
href="mailto:asterisk@gafana.com">asterisk@gafana.com</A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<DIV style="DIRECTION: ltr">
<DIV><FONT face=Arial size=2>Hi Ron,</FONT></DIV>
<DIV><FONT face=Arial size=2> 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.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>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.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>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?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>If I can get my ServerIron working, I will do
a complete write up on it...but it works only partially.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>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?</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>- Gabe</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(0,0,0) 2px solid; MARGIN-RIGHT: 0px"></BLOCKQUOTE></DIV>
<DIV style="DIRECTION: ltr"><SPAN class=q>
<DIV
style="FONT: 10pt arial; font-size-adjust: none; font-stretch: normal">-----
Original Message ----- </DIV>
<DIV
style="BACKGROUND: rgb(228,228,228) 0% 50%; FONT: 10pt arial; font-size-adjust: none; font-stretch: normal; moz-background-clip: initial; moz-background-origin: initial; moz-background-inline-policy: initial"><B>From:</B>
<A title=ronmccar@gmail.com
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:ronmccar@gmail.com" target=_blank>Ron McCarthy</A>
</DIV></SPAN></DIV>
<DIV style="DIRECTION: ltr"><SPAN class=q>
<DIV
style="FONT: 10pt arial; font-size-adjust: none; font-stretch: normal"><B>To:</B>
<A title=asterisk@gafana.com
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:asterisk@gafana.com" target=_blank>Gabriel Afana</A> ; <A
title=asterisk-users@lists.digium.com
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:asterisk-users@lists.digium.com" target=_blank>Asterisk Users
Mailing List -Non-Commercial Discussion</A> </DIV>
<DIV
style="FONT: 10pt arial; font-size-adjust: none; font-stretch: normal"><B>Sent:</B>
Sunday, March 12, 2006 1:16 PM</DIV>
<DIV
style="FONT: 10pt arial; font-size-adjust: none; font-stretch: normal"><B>Subject:</B>
Re: [Asterisk-Users] Clustering</DIV>
<DIV><BR></DIV></SPAN></DIV>
<DIV style="DIRECTION: ltr"><SPAN class=e id=q_109f0782c11f53d8_3>Hi
Gabe,<BR>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?<BR><BR>Thanks!<BR>Ron<BR><BR>
<DIV><SPAN class=gmail_quote>On 3/11/06, <B class=gmail_sendername>Gabriel
Afana</B> <<A onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:asterisk@gafana.com"
target=_blank>asterisk@gafana.com</A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<DIV style="DIRECTION: ltr">
<DIV><FONT face=Arial size=2>Hi Ron,</FONT></DIV>
<DIV><FONT face=Arial size=2> 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.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>- Gabe</FONT></DIV>
<DIV> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(0,0,0) 2px solid; MARGIN-RIGHT: 0px"></BLOCKQUOTE></DIV>
<DIV style="DIRECTION: ltr"><SPAN>
<DIV
style="FONT: 10pt arial; font-size-adjust: none; font-stretch: normal">-----
Original Message ----- </DIV>
<DIV
style="BACKGROUND: rgb(228,228,228) 0% 50%; FONT: 10pt arial; font-size-adjust: none; font-stretch: normal; moz-background-clip: initial; moz-background-origin: initial; moz-background-inline-policy: initial"><B>From:</B>
<A title=ronmccar@gmail.com
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:ronmccar@gmail.com" target=_blank>Ron McCarthy</A>
</DIV></SPAN></DIV>
<DIV style="DIRECTION: ltr"><SPAN>
<DIV
style="FONT: 10pt arial; font-size-adjust: none; font-stretch: normal"><B>To:</B>
<A title=asterisk-users@lists.digium.com
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:asterisk-users@lists.digium.com" target=_blank>Asterisk
Users Mailing List - Non-Commercial Discussion</A> </DIV></SPAN></DIV>
<DIV style="DIRECTION: ltr"><SPAN>
<DIV
style="FONT: 10pt arial; font-size-adjust: none; font-stretch: normal"><B>Sent:</B>
Friday, March 10, 2006 11:22 AM</DIV>
<DIV
style="FONT: 10pt arial; font-size-adjust: none; font-stretch: normal"><B>Subject:</B>
[Asterisk-Users] Clustering</DIV>
<DIV><BR></DIV>Hello All,<BR><BR>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.<BR><BR>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..<BR><BR>Any
comments would be great!<BR><BR>Thanks<BR>Ron<BR></SPAN></DIV>
<DIV style="DIRECTION: ltr">
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