19 Mar
2015
19 Mar
'15
6:54 a.m.
Hi, all
I'm trying to set up 2 kamailio servers for active-active redundancy. The two kamailio
severs share the the same database with db_mode=3, and no registration replication. Use
pjsua2 as SIP client for testing. The test setup is as follows:
kamailio server 1(k1): 10.0.1.30:5061 <http://10.0.1.30:5061/>
kamailio server 2(k2): 10.0.1.32:5061 <http://10.0.1.32:5061/>
sip client c1: sip:16317@10.0.1.30 <mailto:sip%3A16317@10.0.1.30>, client ip:
10.0.1.254
sip client c2: sip:72316@10.0.1.30 <mailto:sip%3A72316@10.0.1.30>, client ip:
10.0.1.254
c1 is registered with k1.
c2 is registered with k2.
When c1 calls c2, the call flow is as follows:
INVITE:
c1------>k1------>k2------>c2
200 OK:
c1<------k1<------k2<------c2
So far, the INVITE works. c1 and c2 can establish a call. However, when c2 hangs up and
sends BYE, the BYE is forwarded all the way to k1, and then k1 gives a 500 server error.
The log on k1 seems to indicate k1 was attempting to create a TLS connection to itself
instead of using the existing TLS connection to c1 to forward the BYE.
BYE:
c1 X k1<------k2<------c2
|
|------->Connection refused/500
Here is the kamailio log on k1:
Mar 16 10:34:21 kamailio.sip.com <http://kamailio.sip.com/>
/usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send():
connect 10.0.1.30:55999 <http://10.0.1.30:55999/> failed (RST) Connection refused
Mar 16 10:34:21 kamailio.sip.com <http://kamailio.sip.com/>
/usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send():
10.0.1.30:55999 <http://10.0.1.30:55999/>: connect & send for 0x7fc96a68a1a0
failed: Connection refused (111)
This is the routing logic for BYE in kamailio.cfg:
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
xlog("L_DBG", "=====BYE $ru from $fu
$si:$sp to $du=====\n");
dlg_manage();
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction
fails
}
else if ( is_method("ACK") ) {
# ACK is forwarded statelessy
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
# Add Record-Route for in-dialog NOTIFY as per RFC 6665.
record_route();
}
route(RELAY);
exit;
} else {......
If I add location based routing for BYE from peer kamailio, then the BYE seems to be
forwarded correctly:
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
xlog("L_DBG", "=====BYE $ru from $fu
$si:$sp to $du=====\n");
dlg_manage();
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction
fails
# If BYE coming from Kamailio peer, route BYE by
location
$var(peerlist)=$sel(cfg_get.trusted.peers);
$var(i) = 0;
while($var(i)<$(var(peerlist){param.count})) {
xlog("L_DBG",
"=====$(var(peerlist){param.count})=====$(var(peerlist){param.valueat,$var(i)})=====\n");
if(src_ip==$(var(peerlist){param.valueat,$var(i)})) {
lookup("location");
xlog("L_DBG", "=====BYE
from $fu $si:$sp to $du=====\n");
break;
}
$var(i) = $var(i) + 1;
}
}
else if ( is_method("ACK") ) {
# ACK is forwarded statelessy
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
# Add Record-Route for in-dialog NOTIFY as per RFC 6665.
record_route();
}
route(RELAY);
exit;
} else {......
My first question is why k1 loose_route sends the BYE to itself instead of the client. Is
this a bug?
My next question is whether the above location routing for BYE from peer kamailio a
good/safe approach.
The SIP traces will be sent later to avoid exceeding email size limit.
Thanks
20 Mar
20 Mar
1 p.m.
New subject: missing BYE when 2 redundant kamailio servers share the same database
Hello,
On 19/03/15 02:54, Ding Ma wrote:
[...]
My first question is why k1 loose_route sends the BYE to itself
instead of the client. Is this a bug?
can you get the pcap of such call? We have to see the routing headers to
say what is next hop address.
Are all the requests within dialog routed via same instance of kamailio?
My next question is whether the above location routing
for BYE from
peer kamailio a good/safe approach.
The SIP traces will be sent later to avoid exceeding email size limit.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
6:52 p.m.
New subject: missing BYE when 2 redundant kamailio servers share the same database
We use TLS for SIP. The Wireshark pcap would be encrypted. I’ll try to get a pcap anyway.
Wonder if there is a way to dump pcap from inside kamailio.
All the requests within dialog are routed through 2 kamailio instances. We want to make
sure each phone only sends requests through its registrar.
I have included pjsua logs in subsequent emails in this thread. Those logs have SIP
messages, but only provide client perspective.
Thanks for the help,
Ding
On Mar 20, 2015, at 3:00 AM, Daniel-Constantin Mierla
<miconda(a)gmail.com> wrote:
Hello,
On 19/03/15 02:54, Ding Ma wrote:
[...]
My first question is why k1 loose_route sends the BYE to itself instead of the client. Is
this a bug?
can you get the pcap of such call? We have to see the routing headers to say what is next
hop address.
Are all the requests within dialog routed via same instance of kamailio?
My next question is whether the above location
routing for BYE from peer kamailio a good/safe approach.
The SIP traces will be sent later to avoid exceeding email size limit.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda <http://twitter.com/#!/miconda> -
http://www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com <http://www.kamailioworld.com/>
7:09 p.m.
New subject: missing BYE when 2 redundant kamailio servers share the same database
We use TLS for SIP. The Wireshark pcap would be
encrypted. I’ll try to
get a pcap anyway. Wonder if there is a way to dump pcap from inside
kamailio.
Wireshark can decrypt SIP signalling sent over TLS connections if you
provide server's private key to it.
All the requests within dialog are routed through 2 kamailio
instances. We want to make sure each phone only sends requests through
its registrar.
I have included pjsua logs in subsequent emails in this thread. Those
logs have SIP messages, but only provide client perspective.
Thanks for the help,
Ding
On Mar 20, 2015, at 3:00 AM, Daniel-Constantin
Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote:
Hello,
On 19/03/15 02:54, Ding Ma wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users [...]
My first question is why k1 loose_route sends the BYE to itself
instead of the client. Is this a bug?
can you get the pcap of such call? We have to see the routing headers
to say what is next hop address.
Are all the requests within dialog routed via same instance of kamailio?
My next question is whether the above location
routing for BYE from
peer kamailio a good/safe approach.
The SIP traces will be sent later to avoid exceeding email size limit.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany -http://www.kamailioworld.com
11:08 p.m.
New subject: missing BYE when 2 redundant kamailio servers share the same database
An alternative would be to test with tcp, from routing point of view
should be the same if it is udp-tcp or udp-tls.
Cheers,
Daniel
On 20/03/15 15:09, Vitaliy Aleksandrov wrote:
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
We use TLS for SIP. The Wireshark pcap would be
encrypted. I’ll try
to get a pcap anyway. Wonder if there is a way to dump pcap from
inside kamailio.
Wireshark can decrypt SIP signalling sent over TLS connections if
you
provide server's private key to it.
All the requests within dialog are routed through 2 kamailio
instances. We want to make sure each phone only sends requests
through its registrar.
I have included pjsua logs in subsequent emails in this thread. Those
logs have SIP messages, but only provide client perspective.
Thanks for the help,
Ding
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users On Mar 20, 2015, at 3:00 AM, Daniel-Constantin
Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote:
Hello,
On 19/03/15 02:54, Ding Ma wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users [...]
My first question is why k1 loose_route sends the BYE to itself
instead of the client. Is this a bug?
can you get the pcap of such call? We have to see the routing
headers to say what is next hop address.
Are all the requests within dialog routed via same instance of kamailio?
My next question is whether the above location
routing for BYE from
peer kamailio a good/safe approach.
The SIP traces will be sent later to avoid exceeding email size limit.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
9 Apr
9 Apr
1:26 a.m.
New subject: missing BYE when 2 redundant kamailio servers share the same database
Hi, all
It took some time to clean up the pjsua settings and collect the pcaps for
both cases.
Here are the two pcap files.
call_failed_bye.pcap has the call trace when the BYE is handled with the
following logic.
route[WITHINDLG] {
if (has_totag()) {
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n");
dlg_manage();
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
}
}
....
route(RELAY);
This generated a 500 error.
call_extra_bye.pcap is the call trace when the BYE is handled like this:
route[WITHINDLG] {
if (has_totag()) {
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n");
dlg_manage();
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
# If BYE is coming from Kamailio peer, route BYE by location
$var(peerlist)=$sel(cfg_get.trusted.peers);
$var(i) = 0;
while($var(i)<$(var(peerlist){param.count})) {
xlog("L_DBG",
"=====$(var(peerlist){param.count})=====$(var(peerlist){param.valueat,$var(i)})=====\n");
if(src_ip==$(var(peerlist){param.valueat,$var(i)})) {
lookup("location");
xlog("L_DBG", "=====BYE from $fu $si:$sp to $du=====\n");
break;
}
$var(i) = $var(i) + 1;
}
}
...
route(RELAY);
This got the BYE and 200 OK for the BYE. But there was an extra BYE at the
end, which resulted in 481 Call/Transaction Does Not Exist.
Wonder if the 481 was caused by pjsua not sending ACK for the 200 OK of BYE.
Just to recap the use case. We're trying to set up two kamailio servers
sharing the same database(db_mode=3).
kamailio server 1(k1): 10.0.1.30:5060
kamailio server 2(k2): 10.0.1.32:5060
sip client c1: sip:16317@10.0.1.30, client ip: 10.0.1.254
sip client c2: sip:72316@10.0.1.30, client ip: 10.0.1.254
c1 is registered with k1.
c2 is registered with k2. The expected call flow is as follows:
When c1 calls c2, the expected call flow is as follows:
INVITE:
c1------>k1------>k2------>c2
200 OK:
c1<------k1<------k2<------c2
BYE
c1<------k1<------k2<------c2
But we have the above-mentioned issue with BYE handling at k1.
Thanks
Ding
On Fri, Mar 20, 2015 at 1:08 PM, Daniel-Constantin Mierla <miconda(a)gmail.com
wrote:
> An alternative would be to test with tcp, from routing point of view
> should be the same if it is udp-tcp or udp-tls.
>
> Cheers,
> Daniel
>
>
> On 20/03/15 15:09, Vitaliy Aleksandrov wrote:
>
>
>
> We use TLS for SIP. The Wireshark pcap would be encrypted. I’ll try to
> get a pcap anyway. Wonder if there is a way to dump pcap from inside
> kamailio.
>
> Wireshark can decrypt SIP signalling sent over TLS connections if you
> provide server's private key to it.
>
>
> All the requests within dialog are routed through 2 kamailio instances. We
> want to make sure each phone only sends requests through its registrar.
>
> I have included pjsua logs in subsequent emails in this thread. Those
> logs have SIP messages, but only provide client perspective.
>
> Thanks for the help,
>
> Ding
>
>
> On Mar 20, 2015, at 3:00 AM, Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
>
> Hello,
>
> On 19/03/15 02:54, Ding Ma wrote:
>
> [...]
>
>
> My first question is why k1 loose_route sends the BYE to itself instead of
> the client. Is this a bug?
>
>
> can you get the pcap of such call? We have to see the routing headers to
> say what is next hop address.
>
> Are all the requests within dialog routed via same instance of kamailio?
>
> My next question is whether the above location routing for BYE from peer
> kamailio a good/safe approach.
>
> The SIP traces will be sent later to avoid exceeding email size limit.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
> Kamailio World Conference, May 27-29, 2015
> Berlin, Germany - http://www.kamailioworld.com
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
> Kamailio World Conference, May 27-29, 2015
> Berlin, Germany - http://www.kamailioworld.com
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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5 comments
3 participants
participants (3)
-
Daniel-Constantin Mierla -
Ding Ma -
Vitaliy Aleksandrov