28 Sep
2009
28 Sep
'09
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
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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
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