[Kamailio-Users] add_path_received and OBP problems
Klaus Darilion
klaus.mailinglists at pernau.at
Fri Sep 25 13:58:34 CEST 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.. at .?.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..... at .@.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.. at .@.. at ..,+.........#>.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.. at .@.. at ..,+.........#>.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.. at .@.. at ..,+.........#>.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.. at .@.. at ..,+.........#>.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..... at .?.\"..,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..... at .@.[...,+..,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.. .. at .@.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..... at .?.[...,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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