25 Sep
2009
25 Sep
'09
4:40 a.m.
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;
}
}
25 Sep
25 Sep
7:58 a.m.
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;
}
}
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
8:07 a.m.
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.
Many thanks ! I did not call loose_route in the LB, I think that will solve it.
Best regards,
Florian
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=[Pr
oxy+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=z9hG4bK3
d51.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=z9hG4bK3
d51.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=z9hG4bK3
d51.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=z9hG4bK3
d51.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=[Pr
oxy+Registrar]
From: "4366480588134"
<sip:4366480588134@[PSTN
Gateway]>;tag=as33339e2e
Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN
Gateway]
To:
<sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec98
25e-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=[Pr
oxy+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=77b8b5897dbef36847302656cec98
25e-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=77b8b5897dbef36847302656cec98
25e-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;
> }
> }
>
>
>
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
2:02 p.m.
Hi,
as thought invite messages are routed now correctly to the public IP of the UAC.
Now I have another problem. ACK and BYE messages have the same problem. The
loadbalancer(s) are trying to send the messages to the private IP, which of course does
not work.
I thought at least the BYE message is an in-dialog-request, and therefore it should work ?
Do you have an idea how to route this kind of messages correctly through the LB to the
UAC's ?
best regards,
Florian
from
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.
Many thanks ! I did not call loose_route in the LB, I think that will
solve it.
Best regards,
Florian
Florian Meister schrieb:
the
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 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.
SIP,
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: length: 1101
SIP,
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: length: 445
[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Pr
E.....@.@.Z...,+..,F........SIP/2.0 100 Giving a
try
Via: SIP/2.0/UDP
oxy+Registrar]
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
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
d51.bf3830b5.0
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
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
d51.bf3830b5.0
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
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
d51.bf3830b5.0
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
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
d51.bf3830b5.0
SIP,
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: length: 397
SIP,
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: length: 413
[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Pr
E.....@.@.[...,+..,F........SIP/2.0 200
canceling
Via: SIP/2.0/UDP
oxy+Registrar]
<sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec98
From: "4366480588134"
<sip:4366480588134@[PSTN
Gateway]>;tag=as33339e2e
Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN
Gateway]
To:
25e-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
[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Pr
E.. ..@.@.Z...,+..,F........SIP/2.0 487
Request Terminated
Via: SIP/2.0/UDP
oxy+Registrar]
<sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec98
Via: SIP/2.0/UDP [PSTN
Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
From: "4366480588134"
<sip:4366480588134@[PSTN
Gateway]>;tag=as33339e2e
To:
25e-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
<sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec98
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:
25e-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 ------------
--
-
"/etc/kamailio/dispatcher.list")
# ----- 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", "$avp(i:80)")
>
> # -- dispatcher params --
> modparam("dispatcher", "list_file",
# -- path params --
modparam("path", "use_received", 1)
####### Routing Logic ########
# main request routing logic
route{
xlog("L_ALERT", "[$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]
[$fu] to [$tu]\n");
[$fu]
t_relay();
exit;
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
xlog("L_ALERT", "_____NON-
LOOSE, STATEFUL-ACK [$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
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;
> }
> }
>
>
>
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
_______________________________________________
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Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
28 Sep
28 Sep
2:34 a.m.
as thought invite messages are routed now correctly to
the public IP of
the UAC.
Now I have another problem. ACK and BYE messages have the same problem.
The loadbalancer(s) are trying to send the messages to the private IP,
which of course does not work.
I thought at least the BYE message is an in-dialog-request, and
therefore it should work ? Do you have an idea how to route this kind
of messages correctly through the LB to the UAC's ?
Hi,
I've solved the problem. No need to answer :D
Best regards, Florian
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.
Many thanks ! I did not call loose_route in the LB, I think that will
solve it.
Best regards,
Florian
>
> Florian Meister schrieb:
> > Hi,
> >
> > I'm trying to set up a HA-OBP for both natted and non-natted > >
> > 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 > 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 > 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
> oxy+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
> d51.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
> d51.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
> d51.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
> d51.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
UAC];user=phone SIP/2.0
SIP,
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: length: 413
E.....@.@.[...,+..,F........SIP/2.0 200
canceling
Via: SIP/2.0/UDP
oxy+Registrar]
From: "4366480588134"
<sip:4366480588134@[PSTN
Gateway]>;tag=as33339e2e
Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN
Gateway]
To:
25e-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
oxy+Registrar]
Via: SIP/2.0/UDP [PSTN
Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
From: "4366480588134"
<sip:4366480588134@[PSTN
Gateway]>;tag=as33339e2e
To:
25e-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:
> 25e-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 ----------
--
--
to
-
"/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 [$fu]
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
#modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
#modparam("rr", "append_fromtag", 0)
# -- nathelper
modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy/rtpproxy.sock")
> modparam("nathelper", "received_avp", "$avp(i:80)")
>
> # -- dispatcher params --
> modparam("dispatcher", "list_file",
[$tu]\n");
> # sequential request withing a dialog should
> # take the path determined by record-routing
> if (loose_route()) {
> xlog("L_ALERT", "_____LOOSE_ROUTE [$rm]
from
> [$fu] to [$tu]\n");
> > t_relay();
> > exit;
> > } else {
> > if ( is_method("ACK") ) {
> > if ( t_check_trans() ) {
> > xlog("L_ALERT",
2:42 a.m.
Florian Meister schrieb:
Hi,
as thought invite messages are routed now correctly to the public IP
of the UAC.
Now I have another problem. ACK and BYE messages have the same
problem. The loadbalancer(s) are trying to send the messages to the
private IP, which of course does not work.
I thought at least the BYE message is an in-dialog-request, and
therefore it should work ? Do you have an idea how to route this kind
of messages correctly through the LB to the UAC's ?
How do you route in-dialog messages? LB and proxy/registrar should use
record-route on initial request to stay in the proxy-chain. LB and
proxy/registrar should route in-dialog requests with loose_route() and
t_relay().
Also, on the LB you have to do the NAT traversal for the clients: i.e.
fix_nated_contact() on messages from the client. If the LB is multihomed
and uses double-record-routing I think you could the same mechanism like
path (adding "received" to the record-route header) too.
regards
klaus
best regards,
Florian
from
_______________________________________________ Kamailio (OpenSER) -
Users mailing list Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
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.
Many thanks ! I did not call loose_route in the LB, I think that
will solve it.
Best regards,
Florian
Florian Meister schrieb:
the
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 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.
SIP,
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: length: 1101
SIP,
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: 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=[Pr
oxy+Registrar]
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
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 d51.bf3830b5.0
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
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 d51.bf3830b5.0
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
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 d51.bf3830b5.0
[Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
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 d51.bf3830b5.0
SIP,
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: length: 397
SIP,
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: length: 413
> E.....@.@.[...,+..,F........SIP/2.0 200 canceling Via:
> SIP/2.0/UDP
[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Pr
oxy+Registrar]
<sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec98
From: "4366480588134"
<sip:4366480588134@[PSTN
Gateway]>;tag=as33339e2e
> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To: 25e-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=[Pr
oxy+Registrar]
<sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec98
Via: SIP/2.0/UDP [PSTN
Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
From: "4366480588134"
<sip:4366480588134@[PSTN
Gateway]>;tag=as33339e2e
> To: 25e-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
<sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec98
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: 25e-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
> ------------
--
-
"/etc/kamailio/dispatcher.list")
# ----- 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", "$avp(i:80)")
>
> # -- dispatcher params -- modparam("dispatcher", "list_file",
# -- path
params -- modparam("path", "use_received", 1)
####### Routing Logic ########
# main request routing logic
route{ xlog("L_ALERT", "[$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] [$fu] to [$tu]\n");
[$fu]
t_relay(); exit; } else { if (
is_method("ACK") ) { if (
t_check_trans() ) { xlog("L_ALERT", "_____NON-
LOOSE, STATEFUL-ACK
[$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 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; } }
>
>
>
>
> _______________________________________________ Kamailio
> (OpenSER) - Users mailing list Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
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3:03 a.m.
How do you route in-dialog messages? LB and
proxy/registrar should use
record-route on initial request to stay in the proxy-chain. LB and
proxy/registrar should route in-dialog requests with loose_route() and
t_relay().
Also, on the LB you have to do the NAT traversal for the clients: i.e.
fix_nated_contact() on messages from the client. If the LB is
multihomed
and uses double-record-routing I think you could the same mechanism
like
path (adding "received" to the record-route header) too.
I already found my problem. It was as you mentioned a NAT problem. In the preloaded-route
(which is added by the proxy for path support) the nat-flag is not set. In the response I
check for the existence of the nat flag for diciding to do fix_nated_contact. Since it
does not exist, I have the wrong contact in the response.
I am checking now for both ";nat=yes" and "received=", because if
there is a received header it is a natted device.
Best regards,
Florian
4:27 a.m.
Florian Meister schrieb:
btw: many providers (and me) choose to apply NAT handling (fixing
contact) in all cases - to simplify the config and security.
regards
klaus
How do you
route in-dialog messages? LB and proxy/registrar should
use record-route on initial request to stay in the proxy-chain. LB
and proxy/registrar should route in-dialog requests with
loose_route() and t_relay().
Also, on the LB you have to do the NAT traversal for the clients:
i.e. fix_nated_contact() on messages from the client. If the LB is
multihomed and uses double-record-routing I think you could the
same mechanism like path (adding "received" to the record-route
header) too.
I already found my problem. It was as you mentioned a NAT problem. In
the preloaded-route (which is added by the proxy for path support)
the nat-flag is not set. In the response I check for the existence of
the nat flag for diciding to do fix_nated_contact. Since it does not
exist, I have the wrong contact in the response.
I am checking now for both ";nat=yes" and "received=", because if
there is a received header it is a natted device. 
5:37 a.m.
On 28.09.2009 10:49 Uhr, Florian Meister wrote:
bandwidth -- all audio is proxied via your server. That may introduce
latency as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla
* Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
* http://www.asipto.com/index.php/sip-router-masterclass/
btw: many
providers (and me) choose to apply NAT handling (fixing
contact) in all cases - to simplify the config and security.
I see. Does that have any negative impacts ?
6:05 a.m.
Florian Meister schrieb:
Depends ....
For SIP NAT traversal. Almost all SIP clients are symmetric
(send/receive SIP on the same port) - thus by always use the "received"
information instead of the provided "Contact" you should not have
negative impacts. Exception: IIRC, I once saw Cisco SIP phones behind a
Cisco PIX which were asymmetric (can't remember if it was SIP or RTP).
Also, applying NAT traversal in all cases, does not allows 3rd party
registrations (which are usually not needed).
Regarding RTP NAT traversal. By activating the media relay for all calls
you have:
- you need more bandwith
- you may need more RTP proxies (high call volume)
- you might add more delay (this is probably a problem if the relay is
in e.g. Europe and caller/callee are in US. If at least on of the
parties is "near" the RTP proxy the additional delay should me negligible)
+ less customers will ask for help due to NAT problems (e.g. NAT
detection does not work, stupid SIP clients ...)
+ some countries require lawful intercept, which must not be
noticeable by the costumer. As the LI requires a media relay (to record
audio) you do not have a choice and have to activate the media relay for
all calls anyway.
regards
klaus
btw: many
providers (and me) choose to apply NAT handling (fixing
contact) in all cases - to simplify the config and security.
I see. Does that have any negative impacts ?
Regards, Florian
_______________________________________________
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Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
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6:18 a.m.
For SIP NAT traversal. Almost all SIP clients are
symmetric
(send/receive SIP on the same port) - thus by always use the "received"
information instead of the provided "Contact" you should not have
negative impacts. Exception: IIRC, I once saw Cisco SIP phones behind a
Cisco PIX which were asymmetric (can't remember if it was SIP or RTP).
I see. So I'll do it the same.
+ some countries require lawful intercept, which
must not be
noticeable by the costumer. As the LI requires a media relay (to record
audio) you do not have a choice and have to activate the media relay
for
all calls anyway.
I have exactly that problem, so I already proxying all rtp-traffic via rtpproxy.
5:05 a.m.
btw. as you can see in this nice drawing, I have two PSTN GW (asterisk-server). Is there a
possibility to tell asterisk to split up the incoming calls to both proxies and react when
one proxy is unreachable ?
thanks,
florian
>>> [UAC] <---> (NAT) <---> [LB]
<---> [REG + Proxy] <---> [PSTN GW]
[LB]
<---> [REG + Proxy] <---> [PSTN GW]
6:10 a.m.
I guess you mean "two proxies"?
Failover in Asterisk is very easy. E.g. Asterisk can forward the call
from PSTN to either proxy 1.1.1.1 or proxy 2.2.2.2.
sip.conf:
[proxy1]
type=peer
host=1.1.1.1
qualify=yes # to detect if proxy is down
[proxy1]
type=peer
host=2.2.2.2
qualify=yes # to detect if proxy is down
extensions.ael|conf:
_0043. => {
// try first proxy
Dial(SIP/${EXTEN}@proxy1);
// if this fails, try second
Dial(SIP/${EXTEN}@proxy2);
}
of course you can add some load-balancing too.
regards
klaus
Florian Meister schrieb:
btw. as you can see in this nice drawing, I have two
PSTN GW
(asterisk-server). Is there a possibility to tell asterisk to split
up the incoming calls to both proxies and react when one proxy is
unreachable ?
thanks,
florian
>>> [UAC] <---> (NAT) <--->
[LB] <---> [REG + Proxy] <---> [PSTN
>>> GW]
[LB] <---> [REG + Proxy] <---> [PSTN GW]
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5599
days inactive
5602
days old
13 comments
3 participants
participants (3)
-
Daniel-Constantin Mierla -
Florian Meister -
Klaus Darilion