[OpenSIPS-Users] v3.1 Active/Active maintain active calls on node failure

[Kevin Wormington]

I’m not using RTPEngine…the upstream proxies are handling all media, NAT traversal, etc. so the OpenSIPS instances can always reach the endpoints.  I’m using clusterer module to share the user location and dialogs with different active tags per node.  There is zero loss of media on switch-over and sometimes a little longer PDD for new calls during switchover until the upstream proxies detect the instance down.   The only part I can’t seem to get to work is handling the final BYE for calls that were on the failed node originally.   The re-invite ping will correct end them but would like to be able to fix it completely…but maybe that is not currently possible.


Thanks,

Kevin
> On Jan 19, 2021, at 9:31 AM, Social Boh via Users <users at lists.opensips.org> wrote:
> 
> To switch calls from one server to another you have to use redis and rptengine using HA with pacemaker y corosync.
> 
> You must have two OpenSIPs, Two RTPEngine, Two Redis servers (primary-replica) Two Mariad  servers (primary/primary)
> 
> With redis you can save calls data (ip, ports, callid) on active server and then use these data on the replica server when swithc to active. On my tests, when switching from a server to another I have between 5 and 10 seconds without audio.
> 
> Regards
> 
> ---
> I'm SoCIaL, MayBe
> 
> El 19/01/2021 a las 10:00 a. m., Kevin Wormington escribió:
>> I’m not using a VIP and I have made some progress by setting a different active tag on each node…then upon node failure setting the failed node's tag to active on remaining node.  This lets the re-invite pinging work, etc.  It’s almost there but the handling of the BYE…they are still sent to the IP of the failed node even after re-invite pings so any in-progress calls from the failed node are zombie when they hang up until the re-invite ping times out (30 seconds).   I found an article about initiating a re-invite on the new node with something like "opensips-cli -x mi dlg_send_sequential callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge body=outbound” but that either seems to terminate the call immediately or say the dialog wasn’t found.
>> 
>> 
>> Thanks,
>> 
>> Kevin
>>> On Jan 19, 2021, at 8:46 AM, Andy Dierlam <adierlam at ptgi-ics.com> wrote:
>>> 
>>> With dialog writing to db that both servers use.   And same tag on both - modparam("dialog", "dlg_sharing_tag", "vip1=active")
>>> had this working on opensips 2.4
>>> 
>>> thanks
>>> Andy
>>> 
>>> 
>>> On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <kworm at missouri-telecom.com> wrote:
>>> Hi,
>>> 
>>> I've been attempting to get a two node active/active setup to work with the v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted by a proxy that handles all of the NAT/media so either OpenSIPS instance can communicate directly with the user.
>>> 
>>> What I have working so far:
>>> 
>>> Registrations and calls work when sent to either node and if you stop OpenSIPS on a node new calls work fine using the other node.
>>> 
>>> What I can’t get to work:
>>> 
>>> Calls that are already in progress to switch between nodes when one node fails.
>>> 
>>> 
>>> I have messed around with various sharing tags…no tag, same tag, different tags but haven’t had any luck.   I’m guessing that I’m missing something to trigger the remaining node to send re-invites.  Has anyone attempted this type of setup and have any ideas?
>>> 
>>> Thanks,
>>> 
>>> Kevin
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