<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2800.1491" name=GENERATOR></HEAD>
<BODY>
<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">
<P></P>
<HR>
</DIV>
<DIV style="DIRECTION: ltr"><SPAN>
<P></P>_______________________________________________<BR>--Bandwidth and
Colocation provided by <A
onclick="return top.js.OpenExtLink(window,event,this)"
href="http://Easynews.com" target=_blank>Easynews.com</A>
--<BR><BR>Asterisk-Users mailing list<BR>To UNSUBSCRIBE or update options
visit:<BR> <A
onclick="return top.js.OpenExtLink(window,event,this)"
href="http://lists.digium.com/mailman/listinfo/asterisk-users"
target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR></SPAN></DIV>
<DIV style="DIRECTION: ltr">
<P></P></DIV></BLOCKQUOTE></DIV><BR></SPAN></DIV>
<DIV
style="DIRECTION: ltr"></DIV></BLOCKQUOTE></DIV><BR></BLOCKQUOTE></BODY></HTML>