[OpenSIPS-Users] Load Balancer as a Stateless Proxy/Registrar
Bobby Smith
bobby.smith at gmail.com
Wed Jun 24 18:50:43 CEST 2009
Greetings,
Having some issues with Opensips 1.5, specifically the load balancer and
dialog modules (and I'm pretty sure they're due to me not understanding
exactly how they're intended to work).
I want to leave the single opensips server as a stateless device that can
act as a registrar. A sip device will register to it, and when a new INVITE
is processed coming from the sip device, the INVITE will be load balanced to
a cluster of application servers depending on parameters (similar to the
asterisk analogy of a conference server, a pstn server, etc etc).
New INVITE requests can come from the application server, which will be
looked up into the location database and forwarded to the correct registered
SIP device.
It's highly desirable that we keep the dialog information consistent between
the application server and the sip devices (for call logging and
troubleshooting purposes) -- in other words, the application server or sip
device is responsible for generating call-id, etc etc etc.
Now, here's where my difficulty is coming into play: When switching to
using the new load balancing module, if I simply use functions in the SL
module, there's no dialog. Obvious, but since the load balancer module
depends on this to know where to send requests to, what happens is that it
looks like everything is working correctly, but if we have for example two
servers in our load balancing pool, ALL requests go to the first server, so
the second server never gets a request.
If I switch this over to stateful (using the TM module), everything works
exactly as intended -- a dialog is created, load balancer knows who has
what, all responses get forwarded accordingly.
So my question is essentially this -- I've read the documentation on the
website posted about loadbalancer in stateful mode, but is there a way to
keep track of the information the module needs to load balance calls in
stateless mode? I can handle routing all requests through things such as RR
and loose routing, I just need it to know that the first server already has
a dialog, and to move on to the next on the list.
I tried using the create_dialog() function on any new initial requests (an
INVITE), but as soon as it's routed and another message comes back (for
example, "ACK"), then we get an error message that no dialog is found for
it. I can send a list_dialogs to the FIFO stack after the invite is sent,
and nothing is listed.
I know in the previous dispatcher module there was a way to hash the call-id
and include a tag on the sip message -- I could see something similar
working here, but would be a little more worried about performance as I
think that would require a database lookup.
Any examples of this module used in stateless mode would be GREATLY
appreciated. Thanks a ton, and I'm loving 1.5 and all the new features!
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