[Freeswitch-users] Clustering FreeSWITCH

Marc Lewis marc at avvatel.com
Wed Oct 29 19:10:32 PDT 2008

I actually spent a big chunk of today doing various tests.  Freeswitch 
is doing all the right things in this scenario.  The problem actually 
turns out to be my router/firewall that I'm testing behind.  When the 
phones register, it only opens up the port back from the IP address of 
the server it registers to, so when the secondary server tries to send 
the invite, the router blocks it.  In case anyone is curious, the 
router/firewall is a Linux box running Arno's Firewall.  I'll be doing 
more tests with different routers to see which ones work and which ones 
don't.  I'll post my results on the wiki page that I'll be creating that 
covers the setup I've got.

For your other points, I'll take them (at least a few of them) one by one.

1.  I'm doing this already to an extent.  My "fs1" box is using a 
floating IP address and is being monitored using Redhat's cluster 
suite.  If that box goes down, the IP's migrate to a backup machine that 
contains identical copies of the configurations and access to the shared 
storage.  While not a load balancer, this keeps the primary switch up 
(except for the wedges that I've been experiencing that I talk about in 
another thread).

The failover switch, my "fs2" box, is running on in a Xen guest machine 
on another server.

2.  Freeswitch can't do what you describe.  I believe that it does have 
the architecture for it, though, and it will just be a SMOP(tm) (Simple 
Matter Of Programming).  Once Freeswitch matures a bit more I expect 
we'll be seeing all sorts of enterprise solutions for it.

3.  True.  Unless you control everything end to end like Cisco's Call 
Manager, you have to deal with what's out there, so you work up 
solutions like the one I've described.

4.  Brian followed up on this point, and he said it better than I could.

5.  I do agree that conferencing needs to be a bit more robust in a 
clustered environment.  However, there is already a lot of that can be 
done to make Freeswitch scale by having multiple boxes and putting 
different conferences on different servers.  Using xml_curl, you can 
write a back-end application that easily routes conferences to multiple 
different boxes to allow some form of load balancing.

6.  I'm not nearly as worried about current calls dropping in the case 
of a failure as I am about new calls being routed and phones being 
registered.  It would be nice in the case of a failure to not have calls 
drop, but not a requirement for me.

7.  Carrierroute works extremely well for me in my environment.  It 
allows me to have great control with least cost routing as well as have 
automatic redundant gateways both in and out.  It also supports the 
shared database model for building in my own redundancies.  The only 
thing that I don't like about it is that I can't selectively handle the 
media path.  With my CR setup it doesn't touch any media at all.  That 
has caused me some issues with one or two of my carriers, but nothing 
that was insurmountable.  The ones I've had problems with expect you to 
be running a b2bua and have media come from the same IP as the SIP 
messages.  For that reason alone I may end up replacing OpenSER with 
Freeswitch at some point in the future and selectively bypass media, but 
only if I can get a configuration as efficient as my CR setup.  If not, 
I'll just add a second Freeswitch gateway that talks only to those 
certain providers.  Not ideal, but it works.

I will be starting a wiki page about everything I've setup within the 
next couple days.

 - Marc

Yuval Hertzog wrote:
> I assume the problem you asked about it happening because the client 
> is disregarding the INVITE from a server with an IP address it was not 
> registered to. If you try to capture the packets going out of your FS 
> (or packets coming in your phone client), I bet you'll see the INVITE 
> request, but no activity thereafter.
> I believe that when considering High-Availability for FreeSWITCH, 
> these issues need to be addressed:
> 1. A shared/floating IP clustering solution such as a load-balancer 
> will only work if the SOFIA hash table is shared between all servers. 
> I donâEUR^(TM)t know if FreeSWITCH entire state is being held in the 
> database or whether some elements are being held in memory.
> 2. FreeSWITCH needs to have shared-bus architecture to allow for a 
> fully clustered solution. Currently, I donâEUR^(TM)t think that two 
> parked channels on different cluster nodes can be bridged in the 
> current architecture because thereâEUR^(TM)s no inter-cluster media 
> switching protocol that I know of.
> 3. A Meshed server approach where different clients are registered to 
> different nodes (like the Cisco Call Manager architecture) seems to be 
> the only immediate option but it is problematic as it requires the 
> client to be configured with a list of redundant servers and most 
> clients donâEUR^(TM)t have that functionality.
> 4. I would strongly recommend reconsidering the use of any DNS feature 
> (such as SRV records) when deploying a telephony infrastructure. Of 
> course, it all depends what this deployment is for. DNS is commonly 
> used in the ITSP space due to the vast audience but enterprises (all 
> sized) are recommended to refrain adding DNS to the list of 
> point-of-failures in their telephony architectures.
> 5. I believe the FreeSWITCH conferencing module needs to be adapted to 
> support clustering in order to scale over more than one server. This 
> is due to the same share-bus issue mentioned earlier.
> 6. In a meshed servers architecture you will need to implement a 
> mechanism that will identify which node in the cluster âEURoeownsâEUR? 
> B-Leg, bridge the call to that node and in that node bridge the call 
> again to B-Leg. When you find a way to implement it (I believe 
> FreeSWITCH to have the tools to enable you to do it now), it would 
> solve your current issue.
> 7. I still have doubts about using carrierroute module opposed to the 
> DISPATCHER module for inbound traffic, mainly because of the 
> registration issue, but I donâEUR^(TM)t have sufficient experience to 
> determine that.
> Anyway, itâEUR^(TM)s very interesting and I definitely like to know 
> how youâEUR^(TM)re going with it.
> *On Thu Oct 30 2:04 , "Anthony Minessale" sent:
> *
>     This all seems right and would make a great wiki page.
>     What you have described *should* work.
>     when a phone registers try doing
>     sofia_contact <user at domain.com
>     <javascript:top.opencompose('user at domain.com','','','')>>
>     from the cli on each box and see what you get.
>     you can also use this function in the dialpan
>     ${sofia_contact(user at domain.com
>     <javascript:top.opencompose('user at domain.com','','','')>)}
>     check that they are both using the same domain name as the profile
>     name
>     or at least have an alais for it etc.
>     if it's a bug i can fix it pretty fast as that is the intended
>     behaviour
>     perhaps you can join irc and get us in the box(s) to have a look
>     at it as we
>     do not have that situation labbed up anywhere.
>     On Tue, Oct 28, 2008 at 6:41 PM, Marc Lewis <marc at avvatel.com
>     <javascript:top.opencompose('marc at avvatel.com','','','')>> wrote:
>         I am in the process of making my FreeSWITCH installation highly
>         available and I'm running into a couple of snags that was
>         hoping that
>         someone may have some insight on.
>         First, the setup as it is now.
>         There are two installations of FS on two different servers,
>         lets call
>         them fs1 and fs2.  They each pull their configurations, dialplan,
>         directory and post CDR's all using mod_curl from a central web
>         server.
>         That part works great.
>         Calls into and out of FS go through an OpenSER proxy set up using
>         carrierroute.  That part also works great for outbound calls
>         to the
>         PSTN.  Inbound calls also come in through this OpenSER proxy
>         and get
>         routed to the primary switch fs1.  That also works perfectly
>         as long as
>         its going to fs1.
>         fs1 and fs2 are both setup to use an ODBC connection to store
>         registrations.  This is pointed to a MySQL database made highly
>         available using the RedHat Cluster Suite on a shared fibre channel
>         partition.  fs1 and fs2 both share the same database.
>          Voicemail storage
>         on fs1 is directly mounted on a GFS2 partition, fs2 is
>         mounting the
>         shared storage from a different server via NFS for no single
>         point of
>         failure.
>         For the phones, I have them setup to use SRV records and have
>         fs1 at
>         priority 10 and fs2 at priority 20 for acme.domain.com
>         <http://acme.domain.com>.  I've tested
>         this and phones register to the correct server and the
>         sip_registration
>         table shows either fs1 or fs2 as the hostname as I would expect.
>         Here is the problem.  If user 100 at acme.domain.com
>         <javascript:top.opencompose('100 at acme.domain.com','','','')>
>         registers on fs2 and a
>         call comes in from the OpenSER proxy to fs1, bridging the call to
>         /sofia/internal/100%acme.domain.com <http://acme.domain.com>
>         from fs1 doesn't ring the phone.  Is
>         there a difference between 'sofia/internal/100%acme.domain.com
>         <http://acme.domain.com>' and
>         'user/100 at acme.domain.com
>         <javascript:top.opencompose('100 at acme.domain.com','','','')>'?
>         Calls out from either fs1 or fs2 routed to the proxy work
>         fine, its just
>         calls coming in from the proxy.  If the call doesn't go to the
>         switch
>         the user is registered on, the user's phone doesn't ring.  It
>         still goes
>         to voicemail, etc., so that part works.
>         Is there a better way to cluster FreeSWITCH than DNS SRV
>         records and a
>         shared state database?
>         Also, as a side note to Anthony, Brian, et al, if this is the
>         best way,
>         I'll be happy to write up a wiki page on how I have this setup
>         with a
>         lot more detail than this.  I was not able to find much in the
>         way of
>         highly available configurations or cluster configurations, so
>         I put
>         together this system using information cobbled from the wiki,
>         mailing
>         list messages and lurking on IRC.
>         Thanks.
>          - Marc
>         --
>         Marc Lewis
>         Avvatel Corporation
>         _______________________________________________
>         Freeswitch-users mailing list
>         Freeswitch-users at lists.freeswitch.org
>         <javascript:top.opencompose('Freeswitch-users at lists.freeswitch.org','','','')>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>     -- 
>     Anthony Minessale II
>     FreeSWITCH http://www.freeswitch.org/
>     ClueCon http://www.cluecon.com/
>     AIM: anthm
>     MSN:anthony_minessale at hotmail.com
>     <javascript:top.opencompose('MSN:anthony_minessale at hotmail.com','','','')>
>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <javascript:top.opencompose('PAYPAL:anthony.minessale at gmail.com','','','')>
>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>     FreeSWITCH Developer Conference
>     sip:888 at conference.freeswitch.org
>     <javascript:top.opencompose('sip:888 at conference.freeswitch.org','','','')>
>     iax:guest at conference.freeswitch.org/888
>     <http://iax:guest@conference.freeswitch.org/888>
>     googletalk:conf+888 at conference.freeswitch.org
>     <javascript:top.opencompose('googletalk:conf_PLUS_888 at conference.freeswitch.org','','','')>
>     pstn:213-799-1400
> ------------------------------------------------------------------------
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Marc Lewis
Avvatel Corporation

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