How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?
26 Feb
2014
26 Feb
'14
12:31 a.m.
As part of a project, I have installed a CentOS 6 test system (a virtual machine) with
Asterisk 11.7.0 and Kamailio 4.1.1 downloaded from
http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_….
I am trying to setup a
combination of Kamailio and Asterisk that will route SIP calls between all the configured
networks in the test setup, in addition to being capable of using Asterisk in order to
handle PSTN and IAX2 calls.
I am using the following online guide to modify my kamailio.cfg:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . Based on
this, I generated the attached patch for my Kamailio configuration
My test setup has the following network interfaces:
eth0: 10.1.0.3, on network 10.1.0.0/24
eth1: 192.168.5.18, on network 192.168.0.0/16
eth2: 10.0.0.2, on network 10.0.0.0/24
lo: 127.0.0.1, on network 127.0.0.0/8
I first configured Asterisk with SIP realtime support (with no Kamailio), and tested that
all configured accounts could register from all interfaces, and that Asterisk could
properly route media between any two disjoint networks. After installing Kamailio,
the guide called for disabling Asterisk SIP authentication by setting passwords to NULL,
and moving Asterisk SIP to a different port (I chose 5080) so that Asterisk and Kamailio
could run on the same machine. At this point, the SIP clients (one softphone
and one VoIP phone) can now register at port 5080 without authentication.
In the process of changing my Kamailio configuration according to the attached patch, the
guide says that I should configure the IP of the network interface as the value of
asterisk.bindip and kamailio.bindip. After performing all required changes,
Kamailio does take over authentication at the default port of 5060. Testing shows that for
all SIP clients with IPs belonging to the same network as the configured asterisk.bindip,
both registration and media exchange work correctly, and that the SIP
clients are still capable of calling into the Asterisk dialplan, and therefore, routing
into Asterisk resources.
For SIP clients in disjoint networks, the failure mode depends on whether mhomed is
enabled or disabled in kamailio.cfg.
For mhomed=0 (or unset), I have the following situation between the two SIP clients (one
at 10.1.0.1, the other at 10.0.0.3), as shown by "sip show peers" in Asterisk
(when asterisk.bindip is set to 192.168.5.18):
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Name/username Host Dyn Forcerport ACL Port Status Description
Realtime
gatitoscomx64am_100/gatit 10.1.0.3 D N A 5060 OK (16 ms)
Cached RT
gatitoscomx64am_101/gatit 10.0.0.2 D N A 5060 OK (36 ms)
Cached RT
gatitoscomx64am_IM101 (Unspecified) D N A 0 UNREACHABLE
Cached RT
3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline]
If I try to call from one SIP client to an extension in the Asterisk dialplan that does
NOT map to a SIP client in a disjoint network, the media exchange works (with negotiatied
media IP in the same network as the SIP client), regardless of whether the
calling client belongs in the same network as asterisk.bindip. If I try to call from the
same SIP client to an extension that maps to a SIP client in a disjoint network, the call
fails, and I get the spoken message about the user at extension such-and-such
being unavailable. Additionally, I get the following error message in the Asterisk logs:
[Feb 25 16:53:14] NOTICE[13807][C-00000003] chan_sip.c: Call from
'gatitoscomx64am_101' (10.0.0.2:5060) to extension 'gatitoscomx64am_101'
rejected because extension not found in context 'gatitoscomx64am-from-internal'.
For mhomed=1, the output of "sip show peers" changes to the following (when
asterisk.bindip is set to 192.168.5.18):
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Name/username Host Dyn Forcerport ACL Port Status Description
Realtime
gatitoscomx64am_100/gatit 192.168.5.18 D N A 5060 OK (19 ms)
Cached RT
gatitoscomx64am_101/gatit 192.168.5.18 D N A 5060 OK (34 ms)
Cached RT
gatitoscomx64am_IM101 (Unspecified) D N A 0 UNREACHABLE
Cached RT
3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline]
From wireshark sniffing, I can see that the SDP payload sent from the client to Kamailio
contains the IP address of the client, which is accessible by both Kamailio and Asterisk.
However, the SDP payload in the OK response sent back to the client contains
a media port with the IP address of asterisk.bindip (the one that appears in the
"Host" column in the "sip show peers" report), not the IP address of
the interface that received the INVITE. This results in broken media negotiation for all
SIP clients
belonging to networks other than the one that contains asterisk.bindip.
In either case, I have to hardcode an IP address in kamailio.cfg, which is not
satisfactory. IPs assigned to interfaces can and do change, especially if the interface is
managed with DHCP. To escape this, I tried setting asterisk.bindip to 127.0.0.1, but
since apparently localhost is also a disjoint network, all of the above described problems
apply.
Related to these issues, I am not satisfied with leaving Asterisk running unauthenticated
SIP at the nonstandard port. Somebody suggested blocking the port with iptables, but I do
not want to rely on this alone. I tried setting bindaddr=127.0.0.1 so that
only Kamailio gets to talk to Asterisk, but this also has the side effect of restricting
the media negotiation to localhost only.
I am asking for help in building a Kamailio/Asterisk configuration that will support all
of the networks and route media between all of them, just as if Asterisk were the only
program running. Ideally, the configuration should not encode the current IP of
any interface (except, maybe, localhost). What is the official name (if any) for the setup
I am describing above? Does it have a standard setup procedure? How is Asterisk secured so
that clients cannot bypass authentication using the Asterisk SIP port
directly?
26 Feb
26 Feb
12:25 p.m.
Puh, too many questions in one email.
First, you should describe what you want to achieve. Eg. is there
routing between the networks done by the server? E.g. can a clinet on
10.1.0.0/24 ping a client on 192.168.0.0/16? If yes, there is no need
for Kamailio/Asterisk to listen on multiple interfaces.
If there is no routing, you need to have a media relay too. Either use
rtpproxy or just configure Asterisk with "canreinvite=no" to avoid media
offloading.
So, what setup have you choosen? Then we can think about problems.
regards
Klaus
Am 25.02.2014 23:31, schrieb Alex Villacís Lasso:
As part of a project, I have installed a CentOS 6 test
system (a virtual
machine) with Asterisk 11.7.0 and Kamailio 4.1.1 downloaded from
http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_….
I am trying to setup a combination of Kamailio and Asterisk that will
route SIP calls between all the configured networks in the test setup,
in addition to being capable of using Asterisk in order to handle PSTN
and IAX2 calls.
I am using the following online guide to modify my kamailio.cfg:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
. Based on this, I generated the attached patch for my Kamailio
configuration
My test setup has the following network interfaces:
eth0: 10.1.0.3, on network 10.1.0.0/24
eth1: 192.168.5.18, on network 192.168.0.0/16
eth2: 10.0.0.2, on network 10.0.0.0/24
lo: 127.0.0.1, on network 127.0.0.0/8
I first configured Asterisk with SIP realtime support (with no
Kamailio), and tested that all configured accounts could register from
all interfaces, and that Asterisk could properly route media between any
two disjoint networks. After installing Kamailio, the guide called for
disabling Asterisk SIP authentication by setting passwords to NULL, and
moving Asterisk SIP to a different port (I chose 5080) so that Asterisk
and Kamailio could run on the same machine. At this point, the SIP
clients (one softphone and one VoIP phone) can now register at port 5080
without authentication.
In the process of changing my Kamailio configuration according to the
attached patch, the guide says that I should configure the IP of the
network interface as the value of asterisk.bindip and kamailio.bindip.
After performing all required changes, Kamailio does take over
authentication at the default port of 5060. Testing shows that for all
SIP clients with IPs belonging to the same network as the configured
asterisk.bindip, both registration and media exchange work correctly,
and that the SIP clients are still capable of calling into the Asterisk
dialplan, and therefore, routing into Asterisk resources.
For SIP clients in disjoint networks, the failure mode depends on
whether mhomed is enabled or disabled in kamailio.cfg.
For mhomed=0 (or unset), I have the following situation between the two
SIP clients (one at 10.1.0.1, the other at 10.0.0.3), as shown by "sip
show peers" in Asterisk (when asterisk.bindip is set to 192.168.5.18):
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Name/username Host Dyn Forcerport ACL Port Status
Description Realtime
gatitoscomx64am_100/gatit 10.1.0.3 D N A 5060 OK (16
ms) Cached RT
gatitoscomx64am_101/gatit 10.0.0.2 D N A 5060 OK (36
ms) Cached RT
gatitoscomx64am_IM101 (Unspecified) D N A 0
UNREACHABLE Cached RT
3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0
offline]
If I try to call from one SIP client to an extension in the Asterisk
dialplan that does NOT map to a SIP client in a disjoint network, the
media exchange works (with negotiatied media IP in the same network as
the SIP client), regardless of whether the calling client belongs in the
same network as asterisk.bindip. If I try to call from the same SIP
client to an extension that maps to a SIP client in a disjoint network,
the call fails, and I get the spoken message about the user at extension
such-and-such being unavailable. Additionally, I get the following error
message in the Asterisk logs:
[Feb 25 16:53:14] NOTICE[13807][C-00000003] chan_sip.c: Call from
'gatitoscomx64am_101' (10.0.0.2:5060) to extension 'gatitoscomx64am_101'
rejected because extension not found in context
'gatitoscomx64am-from-internal'.
For mhomed=1, the output of "sip show peers" changes to the following
(when asterisk.bindip is set to 192.168.5.18):
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Name/username Host Dyn Forcerport ACL Port Status
Description Realtime
gatitoscomx64am_100/gatit 192.168.5.18 D N A 5060 OK (19
ms) Cached RT
gatitoscomx64am_101/gatit 192.168.5.18 D N A 5060 OK (34
ms) Cached RT
gatitoscomx64am_IM101 (Unspecified) D N A 0
UNREACHABLE Cached RT
3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0
offline]
From wireshark sniffing, I can see that the SDP payload sent from the
client to Kamailio contains the IP address of the client, which is
accessible by both Kamailio and Asterisk. However, the SDP payload in
the OK response sent back to the client contains a media port with the
IP address of asterisk.bindip (the one that appears in the "Host" column
in the "sip show peers" report), not the IP address of the interface
that received the INVITE. This results in broken media negotiation for
all SIP clients belonging to networks other than the one that contains
asterisk.bindip.
In either case, I have to hardcode an IP address in kamailio.cfg, which
is not satisfactory. IPs assigned to interfaces can and do change,
especially if the interface is managed with DHCP. To escape this, I
tried setting asterisk.bindip to 127.0.0.1, but since apparently
localhost is also a disjoint network, all of the above described
problems apply.
Related to these issues, I am not satisfied with leaving Asterisk
running unauthenticated SIP at the nonstandard port. Somebody suggested
blocking the port with iptables, but I do not want to rely on this
alone. I tried setting bindaddr=127.0.0.1 so that only Kamailio gets to
talk to Asterisk, but this also has the side effect of restricting the
media negotiation to localhost only.
I am asking for help in building a Kamailio/Asterisk configuration that
will support all of the networks and route media between all of them,
just as if Asterisk were the only program running. Ideally, the
configuration should not encode the current IP of any interface (except,
maybe, localhost). What is the official name (if any) for the setup I am
describing above? Does it have a standard setup procedure? How is
Asterisk secured so that clients cannot bypass authentication using the
Asterisk SIP port directly?
_______________________________________________
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
6:39 p.m.
El 26/02/14 05:25, Klaus Darilion escribió:
Puh, too many questions in one email.
I am sorry about that. I wanted to provide as much useful information as possible
on my email, including what I had observed, and what I have already tried. After all the
exposition, I just want to 1) have a setup that routes media between several disjoint
networks, 2) while being secure by default even if the firewall gets misconfigured, 3)
without having to hardcode a particular IP address in the configuration file.
First, you should describe what you want to achieve.
Eg. is there routing between the networks done by the server? E.g. can a clinet on
10.1.0.0/24 ping a client on 192.168.0.0/16? If yes, there is no need for
Kamailio/Asterisk to listen on multiple
interfaces.
No, the server is not supposed to route ordinary traffic between networks.
If there is no routing, you need to have a media relay
too. Either use rtpproxy or just configure Asterisk with "canreinvite=no" to
avoid media offloading.
The asterisk instance was supposed to be already configured with directmedia=no the
whole time. This was set both in the configuration file and in each of the realtime peers.
The full sip.conf file is attached. The #include'd files at the end are currently
empty. You can see the commented-out attempt to listen on localhost only.
I fail to see why rtpproxy would be required in this setup, since asterisk already opens
RTP ports on all interfaces. In the (already tested) asterisk-only scenario, media is
routed between the test networks through asterisk when it does all of the SIP
negotiation itself.
So, what setup have you choosen? Then we can think
about problems.
regards
Klaus
Am 25.02.2014 23:31, schrieb Alex Villacís Lasso:
As part of a project, I have installed a CentOS 6
test system (a virtual
machine) with Asterisk 11.7.0 and Kamailio 4.1.1 downloaded from
http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_….
I am trying to setup a combination of Kamailio and Asterisk that will
route SIP calls between all the configured networks in the test setup,
in addition to being capable of using Asterisk in order to handle PSTN
and IAX2 calls.
I am using the following online guide to modify my kamailio.cfg:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
. Based on this, I generated the attached patch for my Kamailio
configuration
My test setup has the following network interfaces:
eth0: 10.1.0.3, on network 10.1.0.0/24
eth1: 192.168.5.18, on network 192.168.0.0/16
eth2: 10.0.0.2, on network 10.0.0.0/24
lo: 127.0.0.1, on network 127.0.0.0/8
I first configured Asterisk with SIP realtime support (with no
Kamailio), and tested that all configured accounts could register from
all interfaces, and that Asterisk could properly route media between any
two disjoint networks. After installing Kamailio, the guide called for
disabling Asterisk SIP authentication by setting passwords to NULL, and
moving Asterisk SIP to a different port (I chose 5080) so that Asterisk
and Kamailio could run on the same machine. At this point, the SIP
clients (one softphone and one VoIP phone) can now register at port 5080
without authentication.
In the process of changing my Kamailio configuration according to the
attached patch, the guide says that I should configure the IP of the
network interface as the value of asterisk.bindip and kamailio.bindip.
After performing all required changes, Kamailio does take over
authentication at the default port of 5060. Testing shows that for all
SIP clients with IPs belonging to the same network as the configured
asterisk.bindip, both registration and media exchange work correctly,
and that the SIP clients are still capable of calling into the Asterisk
dialplan, and therefore, routing into Asterisk resources.
For SIP clients in disjoint networks, the failure mode depends on
whether mhomed is enabled or disabled in kamailio.cfg.
For mhomed=0 (or unset), I have the following situation between the two
SIP clients (one at 10.1.0.1, the other at 10.0.0.3), as shown by "sip
show peers" in Asterisk (when asterisk.bindip is set to 192.168.5.18):
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Name/username Host Dyn Forcerport ACL Port Status
Description Realtime
gatitoscomx64am_100/gatit 10.1.0.3 D N A 5060 OK (16
ms) Cached RT
gatitoscomx64am_101/gatit 10.0.0.2 D N A 5060 OK (36
ms) Cached RT
gatitoscomx64am_IM101 (Unspecified) D N A 0
UNREACHABLE Cached RT
3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0
offline]
If I try to call from one SIP client to an extension in the Asterisk
dialplan that does NOT map to a SIP client in a disjoint network, the
media exchange works (with negotiatied media IP in the same network as
the SIP client), regardless of whether the calling client belongs in the
same network as asterisk.bindip. If I try to call from the same SIP
client to an extension that maps to a SIP client in a disjoint network,
the call fails, and I get the spoken message about the user at extension
such-and-such being unavailable. Additionally, I get the following error
message in the Asterisk logs:
[Feb 25 16:53:14] NOTICE[13807][C-00000003] chan_sip.c: Call from
'gatitoscomx64am_101' (10.0.0.2:5060) to extension 'gatitoscomx64am_101'
rejected because extension not found in context
'gatitoscomx64am-from-internal'.
For mhomed=1, the output of "sip show peers" changes to the following
(when asterisk.bindip is set to 192.168.5.18):
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Name/username Host Dyn Forcerport ACL Port Status
Description Realtime
gatitoscomx64am_100/gatit 192.168.5.18 D N A 5060 OK (19
ms) Cached RT
gatitoscomx64am_101/gatit 192.168.5.18 D N A 5060 OK (34
ms) Cached RT
gatitoscomx64am_IM101 (Unspecified) D N A 0
UNREACHABLE Cached RT
3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0
offline]
From wireshark sniffing, I can see that the SDP payload sent from the
client to Kamailio contains the IP address of the client, which is
accessible by both Kamailio and Asterisk. However, the SDP payload in
the OK response sent back to the client contains a media port with the
IP address of asterisk.bindip (the one that appears in the "Host" column
in the "sip show peers" report), not the IP address of the interface
that received the INVITE. This results in broken media negotiation for
all SIP clients belonging to networks other than the one that contains
asterisk.bindip.
In either case, I have to hardcode an IP address in kamailio.cfg, which
is not satisfactory. IPs assigned to interfaces can and do change,
especially if the interface is managed with DHCP. To escape this, I
tried setting asterisk.bindip to 127.0.0.1, but since apparently
localhost is also a disjoint network, all of the above described
problems apply.
Related to these issues, I am not satisfied with leaving Asterisk
running unauthenticated SIP at the nonstandard port. Somebody suggested
blocking the port with iptables, but I do not want to rely on this
alone. I tried setting bindaddr=127.0.0.1 so that only Kamailio gets to
talk to Asterisk, but this also has the side effect of restricting the
media negotiation to localhost only.
I am asking for help in building a Kamailio/Asterisk configuration that
will support all of the networks and route media between all of them,
just as if Asterisk were the only program running. Ideally, the
configuration should not encode the current IP of any interface (except,
maybe, localhost). What is the official name (if any) for the setup I am
describing above? Does it have a standard setup procedure? How is
Asterisk secured so that clients cannot bypass authentication using the
Asterisk SIP port directly?
_______________________________________________
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
_______________________________________________
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
28 Feb
28 Feb
12:43 a.m.
El 26/02/14 11:39, Alex Villacís Lasso escribió:
El 26/02/14 05:25, Klaus Darilion escribió:
Regarding security, I think rtpproxy can help me. I have configured asterisk to listen for
SIP traffic on localhost only. Of course, this means asterisk will send the media to
localhost only, but now I can be sure there is no way to bypass kamailio
authentication from the outside.
My test setup resembles a typical NAT traversal. However, instead of a single private
interface and a single outgoing interface, I need to route between localhost (which
behaves here as the private network) and multiple external interfaces (similar to the
"public" network). A partially-working patch to kamailio.cfg is attached to this
email. The known problems with the patch are:
- Incoming calls (external->kamailio->asterisk) are handled and media is correctly
routed with multiple rtpproxy instances. However, calls generated from Asterisk itself
(asterisk->kamailio->external) do not have their media routed correctly. This is
because I need to choose which rtpproxy set to use in order to route the media, and I do
not know which field to use at the required point for an outgoing INVITE. For incoming
INVITEs, I am using $Ri, and for OK replies to those INVITES, I am using
$(hdr(Record-Route)[1] (I believe this is not optimal and that a better solution must
exist).
- I am forced to hardcode IPs to the various defined interfaces. As explained before, IPs
can and do change, and removing the need to hardcode the IPs will make configuration and
maintenance easier. I must add that I want to build a generic solution that
can be minimally modified for several deployments.
Is this setup recognizable as an already-solved problem (minus the localhost trick)? How
is it done correctly?
Puh, too many questions in one email.
I am sorry about that. I wanted to provide as much useful information as possible
on my email, including what I had observed, and what I have already tried. After all the
exposition, I just want to 1) have a setup that routes media between several
disjoint networks, 2) while being secure by default even if the firewall gets
misconfigured, 3) without having to hardcode a particular IP address in the configuration
file.
First, you should describe what you want to
achieve. Eg. is there routing between the networks done by the server? E.g. can a clinet
on 10.1.0.0/24 ping a client on 192.168.0.0/16? If yes, there is no need for
Kamailio/Asterisk to listen on multiple
interfaces.
No, the server is not supposed to route ordinary traffic between networks.
If there is no routing, you need to have a media
relay too. Either use rtpproxy or just configure Asterisk with "canreinvite=no"
to avoid media offloading.
The asterisk instance was supposed to be already configured with directmedia=no
the whole time. This was set both in the configuration file and in each of the realtime
peers. The full sip.conf file is attached. The #include'd files at the end are
currently empty. You can see the commented-out attempt to listen on localhost only.
I fail to see why rtpproxy would be required in this setup, since asterisk already opens
RTP ports on all interfaces. In the (already tested) asterisk-only scenario, media is
routed between the test networks through asterisk when it does all of the SIP
negotiation itself.
1 Mar
1 Mar
6:54 p.m.
Am 27.02.2014 23:43, schrieb Alex Villacís Lasso:
Is this setup recognizable as an already-solved
problem (minus the
localhost trick)? How is it done correctly?
I think your setup is too complex. If
I didn't missed your requirements
I think you can do it this way:
From routing point of view, you could do the setup with Asterisk,
without KAmailio. IF you need Kamailio too (for whatever features) then
what about this:
| Server |
| |
client1---net1-------| IP1:5060 Kamailio |<-------\
| | \
client2---net2-------| IP2:5060 Kamailio |<--\ |
| | / /
| 0.0.0.0:5070 Asterisk |<-------
| 0.0.0.0:10000-20000: Asterisk RTP |
Block port 5070 from outside with iptables, so clients are forced
through Kamailio. For the communication between Asterisk and Kamailio
use the same socket as for the communication with the clients and
Kamailio, eg:
client1-IP ------- IP1:5060 --- Kamailio ---- IP1:5060
|
IP1:5070 ----
\
Asterisk
/
IP2:5070 ---
|
client2-IP ------- IP2:5060 --- Kamailio ---- IP2:5060
Asterisk usually uses the local routing table to find the IP address to
put in the SDP. As the communication with client 1 is via Kamailio IP1
and with client 2 via Kamailio IP2, Asterisk should populate the SDP
correctly.
Further, if you configure Kamailio with mhomed=1 and forward to ASterisk
always using the proper IP, e.g a request which is received on IP1:5060
is forwarded to IP1:5070, then also the Route/Record-Route and Via
headers should be populated correctly. For REGISTER routing I think it
is best to use Path header (new Asterisks should support this).
regards
Klaus
PS: I have not tried it but I am quite sure it should work.
3925
days inactive
3929
days old
4 comments
2 participants
participants (2)
-
Alex Villacís Lasso -
Klaus Darilion