Re: [Kamailio-Users] add_path_received and OBP problems

25 Sep 2009

The request sent out by the LB still has the Route header. This 
indicates that either:
  - the load balancer does not detect the domain/IP in the Route header 
as an internal one (alias=....)
  - the loose_route function is not called in the LB.

regards
klaus

Florian Meister schrieb:
...
  Hi,

 I'm trying to set up a HA-OBP for both natted and non-natted users.

 Atm my setup looks like that:

 [UAC] <---> (NAT) <---> [LB] <---> [REG + Proxy] <---> [PSTN GW]

 At the loadbalancer I use add_path and add_path_received to add a path-header to the
request sent to the Registrar. (Of course only for register requests). I use a few
parameters for telling the registrar module to use the path header. When looking in the
mysql-db I can see the correct value in the path-field in the location-table.

 When getting an invite from the PSTN gateway for one of the UAC's the Proxy is adding
a route-header according to the path-value in the mysql. That seems also correct. Now to
my problem:

 The loadbalancer is sending out the INVITE to the private-ip, not to the public-nat-ip of
the UAC. It seems that the received-parameter in the Route-header is somehow ignored. I am
using the use_received parameter in the loadbalancer, but it does not work.

 In the following you can find the SIP trace (made on the loadbalancer) and the
loadbalancer config, since I think there is the problem.

 Thanks,

 Florian

 Here is the SIP-trace of the INVITE:

 09:49:07.785294 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 1101
 E..i..@.?.Yb..,F..,+.....U.BINVITE sip:435572949012@[PRIVATE IP of UAC];user=phone
SIP/2.0
 Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e>
 Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0
 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of
UAC]:64224">
 Max-Forwards: 69
 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e
 To: <sip:+435572949012@[Proxy+Registrar]>
 Contact: <sip:4366480588134@[PSTN Gateway]>
 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway]
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.6.1.5
 Date: Fri, 25 Sep 2009 07:40:26 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 312

 v=0
 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway]
 s=Asterisk PBX 1.6.1.5
 c=IN IP4 [PSTN Gateway]
 t=0 0
 m=audio 19178 RTP/AVP 8 3 0 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:3 GSM/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv

 09:49:07.786459 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 445
 E.....@.@.Z...,+..,F........SIP/2.0 100 Giving a try
 Via: SIP/2.0/UDP
[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar]
 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e
 To: <sip:+435572949012@[Proxy+Registrar]>
 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway]
 CSeq: 102 INVITE
 Server: Kamailio (1.5.1-notls (i386/linux))
 Content-Length: 0

 09:49:07.786608 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307
 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone
SIP/2.0
 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes>
 Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e>
 Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0
 Via: SIP/2.0/UDP
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0
 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of
UAC]:64224">
 Max-Forwards: 68
 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e
 To: <sip:+435572949012@[Proxy+Registrar]>
 Contact: <sip:4366480588134@[PSTN Gateway]>
 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway]
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.6.1.5
 Date: Fri, 25 Sep 2009 07:40:26 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 330
 Path: <sip:[Loadbalancer];lr>

 v=0
 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway]
 s=Asterisk PBX 1.6.1.5
 c=IN IP4 [Loadbalancer]
 t=0 0
 m=audio 35480 RTP/AVP 8 3 0 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:3 GSM/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv
 a=nortpproxy:yes

 09:49:08.258116 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307
 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone
SIP/2.0
 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes>
 Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e>
 Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0
 Via: SIP/2.0/UDP
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0
 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of
UAC]:64224">
 Max-Forwards: 68
 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e
 To: <sip:+435572949012@[Proxy+Registrar]>
 Contact: <sip:4366480588134@[PSTN Gateway]>
 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway]
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.6.1.5
 Date: Fri, 25 Sep 2009 07:40:26 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 330
 Path: <sip:[Loadbalancer];lr>

 v=0
 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway]
 s=Asterisk PBX 1.6.1.5
 c=IN IP4 [Loadbalancer]
 t=0 0
 m=audio 35480 RTP/AVP 8 3 0 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:3 GSM/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv
 a=nortpproxy:yes

 09:49:09.258109 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307
 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone
SIP/2.0
 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes>
 Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e>
 Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0
 Via: SIP/2.0/UDP
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0
 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of
UAC]:64224">
 Max-Forwards: 68
 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e
 To: <sip:+435572949012@[Proxy+Registrar]>
 Contact: <sip:4366480588134@[PSTN Gateway]>
 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway]
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.6.1.5
 Date: Fri, 25 Sep 2009 07:40:26 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 330
 Path: <sip:[Loadbalancer];lr>

 v=0
 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway]
 s=Asterisk PBX 1.6.1.5
 c=IN IP4 [Loadbalancer]
 t=0 0
 m=audio 35480 RTP/AVP 8 3 0 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:3 GSM/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv
 a=nortpproxy:yes

 09:49:11.258107 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307
 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone
SIP/2.0
 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes>
 Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e>
 Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0
 Via: SIP/2.0/UDP
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0
 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of
UAC]:64224">
 Max-Forwards: 68
 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e
 To: <sip:+435572949012@[Proxy+Registrar]>
 Contact: <sip:4366480588134@[PSTN Gateway]>
 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway]
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.6.1.5
 Date: Fri, 25 Sep 2009 07:40:26 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 330
 Path: <sip:[Loadbalancer];lr>

 v=0
 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway]
 s=Asterisk PBX 1.6.1.5
 c=IN IP4 [Loadbalancer]
 t=0 0
 m=audio 35480 RTP/AVP 8 3 0 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:3 GSM/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv
 a=nortpproxy:yes

 09:49:15.023388 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 397
 E.....@.?.\"..,F..,+......a.CANCEL sip:435572949012@[PRIVATE IP of UAC];user=phone
SIP/2.0
 Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0
 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e
 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway]
 To: <sip:+435572949012@[Proxy+Registrar]>
 CSeq: 102 CANCEL
 Max-Forwards: 70
 User-Agent: OpenSER (1.3.2-notls (x86_64/linux))
 Content-Length: 0

 09:49:15.024138 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 413
 E.....@.@.[...,+..,F........SIP/2.0 200 canceling
 Via: SIP/2.0/UDP
[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar]
 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e
 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway]
 To:
<sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87
 CSeq: 102 CANCEL
 Server: Kamailio (1.5.1-notls (i386/linux))
 Content-Length: 0

 09:49:15.024576 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 493
 E..     ..@.@.Z...,+..,F........SIP/2.0 487 Request Terminated
 Via: SIP/2.0/UDP
[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar]
 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e
 To:
<sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87
 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway]
 CSeq: 102 INVITE
 Server: Kamailio (1.5.1-notls (i386/linux))
 Content-Length: 0

 09:49:15.025180 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 433
 E.....@.?.[...,F..,+.......MACK sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0
 Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0
 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e
 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway]
 To:
<sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87
 CSeq: 102 ACK
 Max-Forwards: 70
 User-Agent: OpenSER (1.3.2-notls (x86_64/linux))
 Content-Length: 0

 ----------------This is the config of my loadbalancer--------------------:

 ####### Global Parameters #########

 debug=100
 log_stderror=yes
 log_facility=LOG_LOCAL0

 fork=no
 children=4

 disable_tcp=yes

 port=5060

 listen=[Loadbalancer]

 ####### Modules Section ########

 mpath="/usr/lib/kamailio/modules/"

 loadmodule "mi_fifo.so"
 loadmodule "sl.so"
 loadmodule "tm.so"
 loadmodule "rr.so"
 loadmodule "pv.so"
 loadmodule "maxfwd.so"
 loadmodule "textops.so"
 loadmodule "siputils.so"
 loadmodule "xlog.so"
 loadmodule "nathelper.so"
 loadmodule "dispatcher.so"
 loadmodule "path.so"

 # ----------------- setting module-specific parameters ---------------

 # ----- mi_fifo params -----
 modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")

 # ----- rr params -----
 # add value to ;lr param to cope with most of the UAs
 #modparam("rr", "enable_full_lr", 1)
 # do not append from tag to the RR (no need for this script)
 #modparam("rr", "append_fromtag", 0)

 # -- nathelper
 modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy/rtpproxy.sock")
 modparam("nathelper", "received_avp", "$avp(i:80)")

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

 # -- path params --
 modparam("path", "use_received", 1)

 ####### Routing Logic ########

 # main request routing logic

 route{
         xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n");
         if (!mf_process_maxfwd_header("10")) {
                 sl_send_reply("483","Too Many Hops");
                 exit;
         }

         # NAT detection
         route(4);

         if (has_totag()) {
                 xlog("L_ALERT", "_____HASTOTAG [$rm] from [$fu] to
[$tu]\n");
                 # sequential request withing a dialog should
                 # take the path determined by record-routing
                 if (loose_route()) {
                         xlog("L_ALERT", "_____LOOSE_ROUTE [$rm] from [$fu]
to [$tu]\n");
                         t_relay();
                         exit;
                 } else {
                         if ( is_method("ACK") ) {
                                 if ( t_check_trans() ) {
                                         xlog("L_ALERT", "_____NON-LOOSE,
STATEFUL-ACK [$rm] from [$fu] to [$tu]\n");
                                         # non loose-route, but stateful ACK; must be an
ACK after a 487 or e.g. 404 from upstream server
                                         t_relay();
                                         exit;
                                 } else {
                                         xlog("L_ALERT",
"_____ACK_WITHOUT_TRANSACTION [$rm] from [$fu] to [$tu]\n");
                                         # ACK without matching transaction ... ignore and
discard.\n");
                                         exit;
                                 }
                         }
                         sl_send_reply("404","Not here");
                 }
                 exit;
         }

         #initial requests

         # CANCEL processing
         if (is_method("CANCEL"))
         {
                 if (t_check_trans())
                         t_relay();
                 exit;
         }

         t_check_trans();

         # record routing
         if (!is_method("REGISTER|MESSAGE"))
                 xlog("L_ALERT", "_____RECORD_ROUTING [$rm] from [$fu] to
[$tu]\n");
                 record_route();

         #if ($rU==NULL) {
         #       # request with no Username in RURI
         #       sl_send_reply("484","Address Incomplete");
         #       exit;
         #}

         if (check_route_param("nat=yes")) {
                 setbflag(6);
         }
         #if (isflagset(5) || isbflagset(6)) {
                 route(5);
         #}

         /* example how to enable some additional event routes */
         if (is_method("INVITE")) {
                 t_on_reply("1");
                 t_on_failure("1");
         }

         if(!ds_is_from_list()) {
                 ds_select_domain("0", "0");
         }
         if (!t_relay()) {
                 sl_reply_error();
         }
 }

 # Caller NAT detection route
 route[4]{
         force_rport();
         if (nat_uac_test("19")) {
                 if (method=="REGISTER") {
                         add_path_received();
                         fix_nated_register();
                 } else {
                         fix_nated_contact();
                 }
                 setflag(5);
         }
         else {
                 add_path();
         }
         return;
 }

 # RTPProxy control
 route[5] {
         if (is_method("BYE")) {
                 unforce_rtp_proxy();
         } else if (is_method("INVITE")){
                 force_rtp_proxy();
         }
         if (!has_totag()) add_rr_param(";nat=yes");
         return;
 }

 onreply_route[1] {
         xdbg("incoming reply\n");

         #if ((isflagset(5) || isbflagset(6)) &&
status=~"(183)|(2[0-9][0-9])") {
         if (status=~"(183)|(2[0-9][0-9])") {
                 force_rtp_proxy();
         }
         if (isbflagset(6)) {
                 fix_nated_contact();
         }
 }

 failure_route[1] {
         #if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) {
         if (is_method("INVITE")) {
                 unforce_rtp_proxy();
         }

         if (t_was_cancelled()) {
                 exit;
         }
 }

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Re: [Kamailio-Users] add_path_received and OBP problems