[SR-Users] Kamailio Load Balancing, problems ACK and BYE.

6 Dec 2012

Hi everyone,

I am a beginner in using Kamailio. I am currently building a load balancer
for multiple SIP Servers, all of them are Kamailio. The Load balancer can
help end-users REGISTER and INVITE normally, but has problems with ACK and
BYE.

Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK
failed, is transmitted to the SIP Server then stop here, doesn't go back to
Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because
doesn't receive ACK, the Called continuously sent 200 call OK to Caller
(sent successfully) and wait for confirmation ACK (after arrival to the SIP
Server, ACK doesn't come back to Load Balancing). Because the Called party
doesn't receive ACK, the call will be automatically disconnected after 30
seconds.

With BYE have rather, if the Caller hangs up first, BYE is transferred over
SIP Server and returned to Load balancer, but Load Balancing can't sent it
to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side
of called). Caller had picked off the phone, but the Caller is not received
BYE and after 30 seconds the call will automatically disconnect (by ACK
errors). If the called party hangs up first, HUNG UP button of the called
even useless, and I don't see any packet of the Called party's BYE message
in Wireshark.

I use X-Lite and EyeBeam for two end users, the configuration of the SIP
Server is the default configuration of Kamailio.

DISPATHCHER.LIST:

I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy
2. But for capturing packets more convenient, I turn off the Proxy 2 and 3.

# group  sip addresses of your * units

# Kamailio SIP Proxies/Registrars

1 sip:192.168.3.11:5060 

#1 sip:192.168.3.12:5060

#1 sip:192.168.3.13:5060  

Previously I used the Stateless Load Balancing configuration by default, but
could not INVITE:

# -- dispatcher params --

modparam("dispatcher", "list_file",
"/etc/kamailio/dispatcher.list")

modparam("dispatcher", "force_dst", 1)

route{

                if ( !mf_process_maxfwd_header("10") )

                {

                               sl_send_reply("483","To Many Hops");

                               drop();

                };

                ds_select_dst("1", "0");

                record_route();

                forward();

                # t_relay();

                }

Then I changed the following, sent INVITE and setup call successfully, but
met ACK and BYE error above.

        if ( is_method("INVITE")) {

        # Not Dispatch INVITE from SIP Proxies

                   if ( ds_is_from_list("1") ) { 

                                t_relay();

                   } else if ( !ds_is_from_list("1") ) {  

                    route(DISPATCH);

                   }

          } else           {   

         # If not INVITE

                    route(DISPATCH);

                }              

      }

    route[DISPATCH] {

    # dispatch destinations 

       ds_select_dst("1", "0");

       record_route();

       t_relay();

       #forward();   

   }           

Please tell me How to modify in the configuration of Proxy and Load
Balancing?

Thank so much for helping!

Best Regard

Nguyen Anh Tuan,

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

The latest 200 OK from Called after Ringing : LB->Caller

Via: SIP/2.0/UDP
172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55
cf46-1--d87543-;rport=9040

Record-Route: <sip:192.168.3.11;lr;nat=yes>

Record-Route: <sip:192.168.3.10;lr=on>

Contact: <sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132>

To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60

From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e