Hi there,
I have been trying for the last week to configure the load balancing in
OpenSIPS. I am trying to configure a load balancer as per the wiki on the
Freeswitch page -
http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS, I have also
tried the sample on the OpenSIPS site and I am getting the same results.
I am attempting to have all my registrations and Invites proxied to the the
Freeswitch server which will do the call processing. My phone is told to point
to the SIP Domain voip2.siptest.net.au which resolves to OpenSIPS.
This is a quick layout of the path:
Yealink T20 - VOIP Phone - 10.2.0.2 (Private IP)
| (NAT)
Cisco 877 DSL Router (2.2.239.241) (Public IP)
|
INTERNET
|
OpenSIPS (1.1.108.70) (Public IP)
|
FreeSWITCH (1.1.108.68) (Public IP)
|
External VOIP Provider
(Assume 2.2.239.X and 1.1.108.X are public ranges)
I register to the OpenSIPS load balancer using domains in this test case I am
using [email protected] where voip2.siptest.net.au is pointed to
1.1.108.70. The registration appears normal on both the OpenSIPS and
Freeswitch. The destination I am calling is via another SIP provider which is
routed by Freeswitch on the external profile.
The problem I am seeing is that it looks like when a RE-INVITE happens the ACK
gets sent back to 2.2.239.241 instead of being relayed to 1.1.108.70, I can
see the ACK from 2.2.239.241 but OpenSIPS then replies and sends the ACK
message to 2.2.239.241 where it should be seeing that it needs to send it to
1.1.108.68. Freeswitch will then keep sending 200 OK to OpenSIPS and then hang
the call up after 30 seconds as there has been no ACK received. If there is no
RE-Invite then the calls seems to work fine. It only seems to be when a
RE-Invite is sent by the Phone. (I have tried a Siemens Gigaset and get the
same issue). If I register the phone directly to Freeswitch I don't seem to
have these issues.
I have seen this issue mentioned on mailing lists in the past and I have tried
the suggestions but none seem to work for me.
I have provided the following which may assist:
opensips.cfg
http://pastebin.com/hG2GUHWV
SIP Trace
http://pastebin.com/8fahFPj3
OpenSIPS debug
http://pastebin.com/T9ULEeXt
Hopefully someone out there might have some ideas. Any advice would be
appreciated.
Cheers,
Ash.
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