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 <[EMAIL PROTECTED]
    <javascript:top.opencompose('[EMAIL PROTECTED]','','','')>>
    from the cli on each box and see what you get.

    you can also use this function in the dialpan
    ${sofia_contact([EMAIL PROTECTED]
    <javascript:top.opencompose('[EMAIL PROTECTED]','','','')>)}

    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 <[EMAIL PROTECTED]
    <javascript:top.opencompose('[EMAIL PROTECTED]','','','')>> 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 [EMAIL PROTECTED]
        <javascript:top.opencompose('[EMAIL PROTECTED]','','','')>
        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/[EMAIL PROTECTED]
        <javascript:top.opencompose('[EMAIL PROTECTED]','','','')>'?

        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


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-- Anthony Minessale II

    FreeSWITCH http://www.freeswitch.org/
    ClueCon http://www.cluecon.com/

    AIM: anthm
    MSN:[EMAIL PROTECTED]
    <javascript:top.opencompose('MSN:[EMAIL PROTECTED]','','','')>
    GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
    <javascript:top.opencompose('PAYPAL:[EMAIL PROTECTED]','','','')>
    IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch

    FreeSWITCH Developer Conference
    sip:[EMAIL PROTECTED]
    <javascript:top.opencompose('sip:[EMAIL PROTECTED]','','','')>
    iax:[EMAIL PROTECTED]/888
    <http://iax:[EMAIL PROTECTED]/888>
    googletalk:[EMAIL PROTECTED]
    <javascript:top.opencompose('googletalk:[EMAIL PROTECTED]','','','')>
    pstn:213-799-1400


------------------------------------------------------------------------

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--
Marc Lewis
Avvatel Corporation

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