I’ve talked about using the UAC module, but as promised, here’s how we can use the UAC module to send SIP REGISTER requests to another SIP server so we can register to another SIP proxy.
Let’s say we’re using Kamailio to talk to a SIP Trunk that requires us to register with them so they know where to send the calls. We’d need to use Kamailio UAC module to manage SIP Registration with our remote SIP Trunk.
But Kamailio’s a proxy, why are we sending requests from it? A proxy just handles messages, right? Proxies don’t originate messages, it’s true, and Kamailio can be a proxy, but with the UAC module we can use Kamailio as a Client instead of a server. Keep in mind Kamailio is what we tell it to be.
Getting Started
Before we can go spewing registrations out all over the internet we need to start by getting a few things in place;
First of which is configuring UAC module, which is something I covered off in my last post,
Once we’ve got that done we’ll need to tell the UAC module our IP Address for the from address for our Contact field, and the database URL of what we’ve setup.
Once we’ve got a database connection in place and UAC module loaded, then we can configure an entry in the uacreg table in the database, in my example I’m going to be registering to an Asterisk box on 192.168.1.205, so I’ll insert that into my database:
mysql> INSERT INTO `uacreg` VALUES (NULL,'myusername','myusername','192.168.1.205','myusername','192.168.1.205','asterisk','myusername','mypassword','','sip:192.168.1.205:5060',60,0,0);
Note: If you’re using a later version of Kamailio (5.4+) then the DB schema changes and you may want something like this:
Having a look at the fields in our table makes it a bit clearer as to what we’ve got in place, setting flags to 0 will see Kamailio attempt registration. Make sure the auth_proxy is a SIP URI (Starts with sip:) and leave the auth_ha1 password empty as we haven’t calculated it.
After we’ve got our database connection in place, UAC module configured and database entries added, it’s time to put it into play, we’ll use Kamcmd to check it’s status:
kamcmd> uac.reg_reload
kamcmd> uac.reg_dump
Unfortunately from Kamcmd we’re not able to see registration status, but Sngrep will show us what’s going on:
From Sngrep we can see the REGISTRATION going out, the authentication challenge and the 200 OK at the end.
Make sure you’ve got your Realm correct, otherwise you may see an error like this:
RROR: {2 10 REGISTER [email protected]} uac [uac_reg.c:946]: uac_reg_tm_callback(): realms do not match. requested realm: [localhost]
Kamailio is a great SIP proxy, but sometimes you might want to see requests originate from Kamailio.
While this isn’t typical proxy behaviour, RFC definitions of a proxy and technical requirements are often two different things. The UAC module allows us to use Kamailio to act as a User Agent Client instead of just a UAS.
There’s one feature I won’t cover in this post, and that’s initiating and outbound SIP Registration using the UAC module, that will get a post of it’s own in the not to distant future.
You may already be sort of using Kamailio is a UAC, if you’re using Dispatcher and sending SIP Pings, then Kamailio is sending SIP OPTIONS messages to the dispatcher destinations. If you’re using the NAT module and sending Keepalives, then you’re also using Kamailio as a UAC. The only difference is the Dispatcher and NAT Helper modules do this for us, and we’re going to originate our own traffic.
There’s a bit of a catch here, when Kamailio receives a request it follows a set of logic and does something with that request. We’re going to remain constrained by this for our example, just to keep things simple.
So let’s work on an example, if a user on our network dials a call to an emergency services number, we’ll send a text message to my IP phone to let me know who’s dialed the emergency services number.
So to start with we’ll need to load the Kamailio UAC module, using LoadModule as we would with any other module:
loadmodule "uac.so"
If you’re working on the default config file that ships with Kamailio you’ll probably have to change how record routing is handled to support UAC,
modparam("rr", "append_fromtag", 1)
Now we should have UAC support added in Kamailio, I’m going to do a bare bones example of the routing logic below, but obviously if you wanted to put this into practice in real life you’d want to actually route the SIP INVITE to an emergency services destination.
First we’ll need to find if the request is an INVITE with the Request URI to an emergency services number, I’ve programmed this in with the Australian emergency services numbers:
if(is_method("INVITE") && ($rU == "000" or $tU == "112" or $tU == "116")){
#Matches any INVITEs with the Request URI to Address as 000, 112 or 116
xlog("Emergency call from $fU to $rU (Emergency number) CSeq is $cs ");
}
Now calls to 000, 112 or 116 will see the alert apear in Xlog:
07:22:41 voice-dev3 /usr/sbin/kamailio[10765]: ERROR: : Emergency call from Test to 112 (Emergency number)
So next up we need to handle the sending a SIP MESSAGE request to my IP phone on the IP 10.0.1.5 – You’re probably thinking we could use the Registrar module to lookup my registered IP address, and you’re right, but to keep things simple I’m just hardcoding it in.
So to keep our routing neat we’ll send calls to the route route(“EmergencyNotify”); and so the demo works I’ll send back a 200 OK and exit – In real life you’d want to handle this request and forward it onto emergency services.
if(is_method("INVITE") && ($rU == "000" or $tU == "112" or $tU == "116")){
#Matches any INVITEs with the Request URI to Address as 000, 112 or 116
xlog("Emergency call from $fU to $rU (Emergency number) CSeq is $cs ");
route("EmergencyNotify");
#You obviously would want this to route to an emergency services destination...
sl_reply("200", "ok");
exit;
}
if(is_method("INVITE")){
#Matches everything else
xlog("Just a regular call from $fU to $rU");
}
Obviously we need to now create a route called route[“EmergencyNotify”]{ } where we’ll put our UAC logic.
For the UAC module we need to craft the SIP Request we’re going to send; we’re going to be sending a SIP MESSAGE request,
So now we’ve sort of put it all together, when a call comes into an emergency destination, like 000, the route EmergencyNotify is called which sends a SIP MESSAGE request to my IP Phone to alert me.
When a caller dials 000 I can see Kamailio sends a SIP MESSAGE to my IP Phone:
Let’s have a look at how this looks on my IP Phone:
When learning to use Kamailio you might find yourself thinking about if you really want to learn to write a Kamailio configuration file, which is another weird scripting language to learn to achieve a task.
Enter KEMI – Kamailio Embedded Interface. KEMI allows you to abstract the routing logic to another programing language. In layman’s terms this means you can write your routing blocks, like request_route{}, reply_route{}, etc, in languages you already know – like Lua, JavaScript, Ruby – and my favorite – Python!
Why would you use KEMI?
Write in a language you already know;
You don’t need to learn how to do write complex routing logic in Kamailio’s native scripting language, you can instead do it in a language you’re already familiar with, writing your Routing Blocks in another programming language.
Change Routing on the Fly;
By writing the routing logic in KEMI allows you to change your routing blocks without having to restart Kamailio, something you can’t do with the “native” scripting language – This means you can change your routing live.
Note: This isn’t yet in place for all languages – Some still require a restart.
Leverage your prefered language’s libraries;
While Kamailio’s got a huge list of modules to interface with a vast number of different things, the ~200 Kamailio modules don’t compare with the thousands of premade libraries that exist for languages like Python, Ruby, JavaScript, etc.
Prerequisites
We’ll obviously need Kamailio installed, but we’ll also need the programming language we want to leverage setup (fairly obvious).
Configuring Kamailio to talk to KEMI
KEMI only takes care of the routing of SIP messages inside our routing blocks – So we’ve still got the Kamailio cfg file (kamailio.cfg) that we use to bind and setup the service as required, load the modules we want and configure them.
Essentially we need to load the app for the language we use, in this example we’ll use app_python3.so and use that as our Config Engine.
Kamailio is generally thought of as a SIP router, but it can in fact handle Diameter signaling as well.
Everything to do with Diameter in Kamailio relies on the C Diameter Peer and CDP_AVP modules which abstract the handling of Diameter messages, and allow us to handle them sort of like SIP messages.
CDP on it’s own doesn’t actually allow us to send Diameter messages, but it’s relied upon by other modules, like CDP_AVP and many of the Kamailio IMS modules, to handle Diameter signaling.
Before we can start shooting Diameter messages all over the place we’ve first got to configure our Kamailio instance, to bring up other Diameter peers, and learn about their capabilities.
C Diameter Peer (Aka CDP) manages the Diameter connections, the Device Watchdog Request/Answers etc, all in the background.
We’ll need to define our Diameter peers for CDP to use so Kamailio can talk to them. This is done in an XML file which lays out our Diameter peers and all the connection information.
In our Kamailio config we’ll add the following lines:
This will load the CDP modules and instruct Kamailio to pull it’s CDP info from an XML config file at /etc/kamailio/diametercfg.xml
Let’s look at the basic example given when installed:
<?xml version="1.0" encoding="UTF-8"?>
<!--
DiameterPeer Parameters
- FQDN - FQDN of this peer, as it should apper in the Origin-Host AVP
- Realm - Realm of this peer, as it should apper in the Origin-Realm AVP
- Vendor_Id - Default Vendor-Id to appear in the Capabilities Exchange
- Product_Name - Product Name to appear in the Capabilities Exchange
- AcceptUnknownPeers - Whether to accept (1) or deny (0) connections from peers with FQDN
not configured below
- DropUnknownOnDisconnect - Whether to drop (1) or keep (0) and retry connections (until restart)
unknown peers in the list of peers after a disconnection.
- Tc - Value for the RFC3588 Tc timer - default 30 seconds
- Workers - Number of incoming messages processing workers forked processes.
- Queue - Length of queue of tasks for the workers:
- too small and the incoming messages will be blocked too often;
- too large and the senders of incoming messages will have a longer feedback loop to notice that
this Diameter peer is overloaded in processing incoming requests;
- a good choice is to have it about 2 times the number of workers. This will mean that each worker
will have about 2 tasks in the queue to process before new incoming messages will start to block.
- ConnectTimeout - time in seconds to wait for an outbound TCP connection to be established.
- TransactionTimeout - time in seconds after which the transaction timeout callback will be fired,
when using transactional processing.
- SessionsHashSize - size of the hash-table to use for the Diameter sessions. When searching for a
session, the time required for this operation will be that of sequential searching in a list of
NumberOfActiveSessions/SessionsHashSize. So higher the better, yet each hashslot will consume an
extra 2xsizeof(void*) bytes (typically 8 or 16 bytes extra).
- DefaultAuthSessionTimeout - default value to use when there is no Authorization Session Timeout
AVP present.
- MaxAuthSessionTimeout - maximum Authorization Session Timeout as a cut-out measure meant to
enforce session refreshes.
-->
<DiameterPeer
FQDN="pcscf.ims.smilecoms.com"
Realm="ims.smilecoms.com"
Vendor_Id="10415"
Product_Name="CDiameterPeer"
AcceptUnknownPeers="0"
DropUnknownOnDisconnect="1"
Tc="30"
Workers="4"
QueueLength="32"
ConnectTimeout="5"
TransactionTimeout="5"
SessionsHashSize="128"
DefaultAuthSessionTimeout="60"
MaxAuthSessionTimeout="300"
>
<!--
Definition of peers to connect to and accept connections from. For each peer found in here
a dedicated receiver process will be forked. All other unkwnown peers will share a single
receiver. NB: You must have a peer definition for each peer listed in the realm routing section
-->
<Peer FQDN="pcrf1.ims.smilecoms.com" Realm="ims.smilecoms.com" port="3868"/>
<Peer FQDN="pcrf2.ims.smilecoms.com" Realm="ims.smilecoms.com" port="3868"/>
<Peer FQDN="pcrf3.ims.smilecoms.com" Realm="ims.smilecoms.com" port="3868"/>
<Peer FQDN="pcrf4.ims.smilecoms.com" Realm="ims.smilecoms.com" port="3868"/>
<Peer FQDN="pcrf5.ims.smilecoms.com" Realm="ims.smilecoms.com" port="3868"/>
<Peer FQDN="pcrf6.ims.smilecoms.com" Realm="ims.smilecoms.com" port="3868"/>
<!--
Definition of incoming connection acceptors. If no bind is specified, the acceptor will bind
on all available interfaces.
-->
<Acceptor port="3868" />
<Acceptor port="3869" bind="127.0.0.1" />
<Acceptor port="3870" bind="192.168.1.1" />
<!--
Definition of Auth (authorization) and Acct (accounting) supported applications. This
information is sent as part of the Capabilities Exchange procedures on connecting to
peers. If no common application is found, the peers will disconnect. Messages will only
be sent to a peer if that peer actually has declared support for the application id of
the message.
-->
<Acct id="16777216" vendor="10415" />
<Acct id="16777216" vendor="0" />
<Auth id="16777216" vendor="10415"/>
<Auth id="16777216" vendor="0" />
<!--
Supported Vendor IDs - list of values which will be sent in the CER/CEA in the
Supported-Vendor-ID AVPs
-->
<SupportedVendor vendor="10415" />
<!--
Realm routing definition.
Each Realm can have a different table of peers to route towards. In case the Destination
Realm AVP contains a Realm not defined here, the DefaultRoute entries will be used.
Note: In case a message already contains a Destination-Host AVP, Realm Routeing will not be
applied.
Note: Routing will only happen towards connected and application id supporting peers.
The metric is used to order the list of prefered peers, while looking for a connected and
application id supporting peer. In the end, of course, just one peer will be selected.
-->
<Realm name="ims.smilecoms.com">
<Route FQDN="pcrf1.ims.smilecoms.com" metric="3"/>
<Route FQDN="pcrf2.ims.smilecoms.com" metric="5"/>
</Realm>
<Realm name="temp.ims.smilecoms.com">
<Route FQDN="pcrf3.ims.smilecoms.com" metric="7"/>
<Route FQDN="pcrf4.ims.smilecoms.com" metric="11"/>
</Realm>
<DefaultRoute FQDN="pcrf5.ims.smilecoms.com" metric="15"/>
<DefaultRoute FQDN="pcrf6.ims.smilecoms.com" metric="13"/>
</DiameterPeer>
First we need to start by telling CDP about the Diameter peer it’s going to be – we do this in the <DiameterPeer section where we define the FQDN and Diameter Realm we’re going to use, as well as some general configuration parameters.
<Peers are of course, Diameter peers. Defining them here will mean a connection is established to each one, Capabilities exchanged and Watchdog request/responses managed. We define the usage of each Peer further on in the config.
The Acceptor section – fairly obviously – sets the bindings for the addresses and ports we’ll listen on.
Next up we need to define the Diameter applications we support in the <Acct id=” /> and <SupportedVendor> parameters, this can be a little unintuitive as we could list support for every Diameter application here, but unless you’ve got a module that can handle those applications, it’s of no use.
Instead of using Dispatcher to manage sending Diameter requests, CDP handles this for us. CDP keeps track of the Peers status and it’s capabilities, but we can group like Peers together, for example we may have a pool of PCRF NEs, so we can group them together into a <Realm >. Instead of calling a peer directly we can call the realm and CDP will dispatch the request to an up peer inside the realm, similar to Dispatcher Groups.
Finally we can configure a <DefaultRoute> which will be used if we don’t specify the peer or realm the request needs to be sent to. Multiple default routes can exist, differentiated based on preference.
We can check the status of peers using Kamcmd’s cdp.list_peers command which lists the peers, their states and capabilities.
One question that’s not as obvious as it perhaps should be is the different states shown with kamcmd dispatcher.list command;
So what do the flags for state mean?
The first letter in the flag means is the current state, Active (A), Inactive (I) or Disabled (D).
The second letter in the flag means monitor status, Probing (P) meaning actively checked with SIP Options pings, or Not Set (X) denoting the device isn’t actively checked with SIP Options pings.
AP – Actively Probing – SIP OPTIONS are getting a response, routing to this destination is possible, and it’s “Up” for all intents and purposes.
IP – Inactively Probing – Destination is not meeting the threshold of SIP OPTIONS request responses it needs to be considered active. The destination is either down or not responding to all SIP OPTIONS pings. Often this is due to needing X number of positive responses before considering the destination as “Up”.
DX – Disabled & Not Probing – This device is disabled, no SIP OPTIONS are sent.
AX – Active & Not Probing– No SIP OPTIONS are sent to check state, but is is effectively “Up” even though the remote end may not be reachable.
In the third part of the Kamailio 101 series I briefly touched upon pseudovariables, but let’s look into what exactly they are and how we can manipulate them to change headers.
The term “pseudo-variable” is used for special tokens that can be given as parameters to different script functions and they will be replaced with a value before the execution of the function.
You’ve probably seen in any number of the previous Kamailio Bytes posts me use pseudovariables, often in xlog or in if statements, they’re generally short strings prefixed with a $ sign like $fU, $tU, $ua, etc.
When Kamailio gets a SIP message it explodes it into a pile of variables, getting the To URI and putting it into a psudovariable called $tU, etc.
We can update the value of say $tU and then forward the SIP message on, but the To URI will now use our updated value.
When it comes to rewriting caller ID, changing domains, manipulating specific headers etc, pseudovariables is where it mostly happens.
Kamailio allows us to read these variables and for most of them rewrite them – But there’s a catch. We can mess with the headers which could result in our traffic being considered invalid by the next SIP proxy / device in the chain, or we could mess with the routing headers like Route, Via, etc, and find that our responses never get where they need to go.
So be careful! Headers exist for a reason, some are informational for end users, others are functional so other SIP proxies and UACs can know what’s going on.
Rewriting SIP From Username Header (Caller ID)
When Kamailio’s SIP parser receives a SIP request/response it decodes the vast majority of the SIP headers into a variety of pseudovariables, we can then reference these variables we can then reference from our routing logic.
Let’s pause here and go back to the Stateless SIP Proxy Example, as we’ll build directly on that.
Follow the instructions in that post to get your stateless SIP proxy up and running, and we’ll make this simple change:
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
xlog("Received $rm to $ru - Forwarding");
$fU = "Nick Blog Example"; #Set From Username to this value
#Forward to new IP
forward("192.168.1.110");
}
Now when our traffic is proxied the From Username will show “Nick Blog Example” instead of what it previously showed.
We’ve talked about using a few different modules, like a SIP Registrar and Htable, that rely on data stored in Kamailio’s memory, the same is true for all the Stateful proxy discussion last week.
But what if you want to share this data between multiple Kamailio instances? This allows distributing workload and every server having the same information and therefore any server is able to process any request.
This allows memory data to be shared between multiple Kamailio instances (aka “Nodes”), so for example if you are storing data in Htable on one Kamailio box, all the other boxes/nodes in the DMQ pool will have the same HTable data.
Kamailio uses SIP to transfer DMQ messages between DMQ nodes, and DNS to discover DMQ nodes.
For this example we’ll share user location data (usrloc) between Kamailio instances, so we’ll create a very simple setup to store location data:
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
#Enable record_routing so we see the BYE / Re-INVITE etc
if(is_method("REGISTER")){
save("location");
}else{
sl_reply("500", "Nope");
}
}
Now if we register a SIP endpoint we should be able to view it using Kamcmd’s ul.dump call, as we talked about in the Kamailio SIP Registrar tutorial.
Next we’ll setup DMQ to allow this data to be shared to other nodes, so they also have the same userloc data available,
First we’ll begin by binding to an extra port for the DMQ messages to go to, to make it a bit clearer what is normal SIP and what’s DMQ,
So for this we’ll add a new line in the config to listen on port 5090:
/* uncomment and configure the following line if you want Kamailio to
* bind on a specific interface/port/proto (default bind on all available) */
listen=udp:0.0.0.0:5060
listen=tcp:0.0.0.0:5060
listen=udp:0.0.0.0:5090
The server_address means we’re listening on any IP on port 5090. In production you may have an IP set here of a private NIC or something non public facing.
The notification address resolves to 2x A records, one is the IP of this Kamailio instance / node, the other is the IP of the other Kamailio instance / node, I’ve just done this in /etc/hosts
Finally we’ll add some routing logic to handle the DMQ messages coming in on port 5090:
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
if (is_method("KDMQ") && $Rp == 5090)
{
dmq_handle_message();
}
#Enable record_routing so we see the BYE / Re-INVITE etc
if(is_method("REGISTER")){
save("location");
}else{
sl_reply("500", "Nope");
}
}
We’ll put the same config on the other Kamailio instance and restart Kamailio on both.
We can now check the DMQ node status to confirm they’re talking to each other.
We talked a little about the Transaction module and using it for Transaction Stateful SIP Proxy, but it’s worth knowing a bit more about the Transaction Module and the powerful functions it offers.
So today I’ll cover some cool functionality TM offers!
Different Reply Routes
By calling the t_on_reply(); we can specify the reply route to be used for replies in this transaction.
route[RELAY]{
#Use reply route "OurReplyRoute" for responses for this transaction
t_on_reply("OurReplyRoute");
#Relay (aka Forward) the request
t_relay_to_udp("192.168.3.118", "5060");
}
onreply_route[OurReplyRoute] {
#On replies from route[RELAY]
#Check our AVP we set in the initial request
xlog("for $rs response the value of AVP \"state_test_var\" is $avp(state_test_var) ");
#Append a header so we can see this was proxied in the SIP capture
append_hf("X-Proxied: For the reply\r\n");
}
Any responses from the route[RELAY] routing block will go to onreply_route[OurReplyRoute], the beauty of this is it allows you to have multiple reply routes each with their own logic. For example for a call leg to a carrier you may want to preserve CLI, but for a call leg to a customer you may wish to restrict it if that’s the option the user has selected, and you can make these changes / modifications in the reply messages.
Failure Routes
Failure routes allow the transaction module to know to try again if a call fails, for example if no response is received from the destination, send it to a different destination, like a backup.
route[RELAY]{
#Use reply route "OurReplyRoute" for responses for this transaction
t_on_reply("OurReplyRoute");
t_on_failure("OurFailureRoute");
#Relay (aka Forward) the request
t_relay_to_udp("192.168.1.118", "5060");
}
failure_route[OurFailureRoute]{
xlog("At failure route");
t_reply("500", "Remote end never got back to us");
exit;
}
We can build upon this, and try a different destination if the first one fails:
request_route {
#Enable record_routing so we see the BYE / Re-INVITE etc
record_route();
#Handle Registrations in a dumb way so they don't messy our tests
if(is_method("REGISTER")){
sl_reply("200", "ok");
exit;
}
#Append a header so we can see this was proxied in the SIP capture
append_hf("X-Proxied: You betcha\r\n");
if(is_method("INVITE")){
#Createa new AVP called "state_test_var" and set the value to "I remember"
$avp(state_test_var) = "I remember";
}
#Let syslog know we've set the value and check it
xlog("for $rm the value of AVP \"state_test_var\" is $avp(state_test_var) ");
#Send to route[RELAY] routing block
rewritehostport("nonexistentdomain.com");
route(RELAY);
}
route[RELAY]{
#Use reply route "OurReplyRoute" for responses for this transaction
t_on_reply("OurReplyRoute");
t_on_failure("OurFailureRoute");
#Relay (aka Forward) the request
t_relay();
}
failure_route[OurFailureRoute]{
xlog("At failure route");
#t_reply("500", "Remote end never got back to us");
rewritehostport("192.168.3.118");
append_branch();
t_relay();
}
onreply_route[OurReplyRoute] {
#On replies from route[RELAY]
#Check our AVP we set in the initial request
xlog("for $rs response the value of AVP \"state_test_var\" is $avp(state_test_var) ");
#Append a header so we can see this was proxied in the SIP capture
append_hf("X-Proxied: For the reply\r\n");
}
One thing to keep in mind is that there’s lots of definitions of failure, for example if you are sending a call to a carrier and get a 404 response back, you probably want to relay that through to the end user, because that destination isn’t there.
But if you get back a 5xx series response you may consider that to be a failure and select the next carrier for example.
Different conditions / requirements have different definitions of “failures” and so there’s a lot to think about when implementing this, along with timeouts for no replies, TCP session management, etc.
Parallel Forking the Call to Multiple Destinations
Parallel Forking is a fancy way of saying ring multiple destinations at the same time.
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
#Enable record_routing so we see the BYE / Re-INVITE etc
record_route();
#Handle Registrations in a dumb way so they don't messy our tests
if(is_method("REGISTER")){
sl_reply("200", "ok");
exit;
}
#Append a header so we can see this was proxied in the SIP capture
append_hf("X-Proxied: You betcha\r\n");
if(is_method("INVITE")){
#Createa new AVP called "state_test_var" and set the value to "I remember"
$avp(state_test_var) = "I remember";
}
#Let syslog know we've set the value and check it
xlog("for $rm the value of AVP \"state_test_var\" is $avp(state_test_var) ");
#Send to route[RELAY] routing block
route(RELAY);
}
route[RELAY]{
#Use reply route "OurReplyRoute" for responses for this transaction
t_on_reply("OurReplyRoute");
#Append branches for each destination we want to forward to
append_branch("sip:[email protected]");
append_branch("sip:[email protected]");
append_branch("sip:[email protected]");
t_on_failure("OurFailureRoute");
#Relay (aka Forward) the request
t_relay();
}
failure_route[OurFailureRoute]{
xlog("At failure route");
t_reply("500", "All those destinations failed us");
}
onreply_route[OurReplyRoute] {
#On replies from route[RELAY]
#Check our AVP we set in the initial request
xlog("for $rs response the value of AVP \"state_test_var\" is $avp(state_test_var) ");
#Append a header so we can see this was proxied in the SIP capture
append_hf("X-Proxied: For the reply\r\n");
}
Bit of a mess but here we see the initial INVITE being branched to the 3 destinations at the same time (Parallel forking)
Serial Forking / Sequential Forking the calls to Multiple Destinations one after the Other
This could be used to try a series of weighted destinations and only try the next if the preceding one fails:
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
#Enable record_routing so we see the BYE / Re-INVITE etc
record_route();
#Send to route[RELAY] routing block
route(RELAY);
}
route[RELAY]{
#Use reply route "OurReplyRoute" for responses for this transaction
t_on_reply("OurReplyRoute");
append_branch("sip:[email protected]", "0.3");
append_branch("sip:[email protected]", "0.2");
append_branch("sip:[email protected]", "0.1");
t_load_contacts();
t_next_contacts();
t_on_failure("OurFailureRoute");
#Relay (aka Forward) the request
t_relay();
break;
}
failure_route[OurFailureRoute]{
xlog("At failure route - Trying next destination");
t_on_failure("OurFailureRoute");
t_relay();
}
onreply_route[OurReplyRoute] {
#On replies from route[RELAY]
#Append a header so we can see this was proxied in the SIP capture
append_hf("X-Proxied: For the reply\r\n");
}
Again this will try each destination, but one after the other based on the weight we added to each destination in the append_branch()
Here we see each destination being tried sequentially
The 3 different proxies all do the same thing, they all relay SIP messages, so we need a way to determine what state has been saved.
To do this we’ll create a variable (actually an AVP) in the initial request (in our example it’ll be an INVITE), and we’ll reference it when handling a response to make sure we’ve got transactional state.
We’ll also try and reference it in the BYE message, which will fail, as we’re only creating a Transaction Stateful proxy, and the BYE isn’t part of the transaction, but in order to see the BYE we’ll need to enable Record Routing.
Stateless Proof
Before we add any state, let’s create a working stateless proxy, and add see how it doesn’t remember:
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
#Enable record_routing so we see the BYE / Re-INVITE etc
record_route();
#Handle Registrations in a dumb way so they don't messy our tests
if(is_method("REGISTER")){
sl_reply("200", "ok");
exit;
}
#Append a header so we can see this was proxied in the SIP capture
append_hf("X-Proxied: You betcha\r\n");
if(is_method("INVITE")){
#Createa new AVP called "state_test_var" and set the value to "I remember"
$avp(state_test_var) = "I remember";
}
#Let syslog know we've set the value and check it
xlog("for $rm the value of AVP \"state_test_var\" is $avp(state_test_var) ");
#Forard to new IP
forward("192.168.3.118");
}
onreply_route{
#Check our AVP we set in the initial request
xlog("for $rs response the value of AVP \"state_test_var\" is $avp(state_test_var) ");
#Append a header so we can see this was proxied in the SIP capture
append_hf("X-Proxied: For the reply\r\n");
}
Now when we run this and call from any phone other than 192.168.3.118, the SIP INVITE will hit the Proxy, and be forwarded to 192.168.3.118.
Syslog will show the INVITE and us setting the var, but for the replies, the value of AVP $avp(state_test_var) won’t be set, as it’s stateless.
Let’s take a look:
kamailio[2577]: {1 1 INVITE [email protected]} ERROR: : for INVITE the value of AVP "state_test_var" is I remember
kamailio[2575]: {2 1 INVITE [email protected]} ERROR: <script>: for 100 response the value of AVP "state_test_var" is <null>
kamailio[2576]: {2 1 INVITE [email protected]} ERROR: <script>: for 180 response the value of AVP "state_test_var" is <null>
kamailio[2579]: {2 1 INVITE [email protected]} ERROR: <script>: for 200 response the value of AVP "state_test_var" is <null>
kamailio[2580]: {1 1 ACK [email protected]} ERROR: <script>: for ACK the value of AVP "state_test_var" is <null>
kamailio[2581]: {1 2 BYE [email protected]} ERROR: <script>: for BYE the value of AVP "state_test_var" is <null>
We can see after the initial INVITE none of the subsequent replies knew the value of our $avp(state_test_var), so we know the proxy is at this stage – Stateless.
Doing the heavy lifting of our state management is the Transaction Module (aka TM). The Transaction Module deserves a post of it’s own (and it’ll get one).
We’ll load the TM module (loadmodule “tm.so”) and use thet_relay() function instead of the forward() function.
But we’ll need to do a bit of setup around this, we’ll need to create a new route block to call t_relay() from (It’s finicky as to where it can be called from), and we’ll need to create a new reply_route{} to manage the response for this particular dialog.
Let’s take a look at the code:
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
#Enable record_routing so we see the BYE / Re-INVITE etc
record_route();
#Handle Registrations in a dumb way so they don't messy our tests
if(is_method("REGISTER")){
sl_reply("200", "ok");
exit;
}
#Append a header so we can see this was proxied in the SIP capture
append_hf("X-Proxied: You betcha\r\n");
if(is_method("INVITE")){
#Createa new AVP called "state_test_var" and set the value to "I remember"
$avp(state_test_var) = "I remember";
}
#Let syslog know we've set the value and check it
xlog("for $rm the value of AVP \"state_test_var\" is $avp(state_test_var) ");
#Send to route[RELAY] routing block
route(RELAY);
}
route[RELAY]{
#Use reply route "OurReplyRoute" for responses for this transaction
t_on_reply("OurReplyRoute");
#Relay (aka Forward) the request
t_relay_to_udp("192.168.3.118", "5060");
}
onreply_route[OurReplyRoute] {
#On replies from route[RELAY]
#Check our AVP we set in the initial request
xlog("for $rs response the value of AVP \"state_test_var\" is $avp(state_test_var) ");
#Append a header so we can see this was proxied in the SIP capture
append_hf("X-Proxied: For the reply\r\n");
}
So unlike before where we just called forward(); to forward the traffic we’re now calling in a routing block called RELAY.
Inside route[RELAY] we set the routing block that will be used to manage replies for this particular transaction, and then call t_relay_to_udp() to relay the request.
We renamed our onreply_route to onreply_route[OurReplyRoute], as specified in the route[RELAY].
So now let’s make a call (INVITE) and see how it looks in the log:
kamailio[5008]: {1 1 INVITE [email protected]} ERROR: : for INVITE the value of AVP "state_test_var" is I remember
kamailio[5005]: {2 1 INVITE [email protected]} ERROR: <script>: for 100 response the value of AVP "state_test_var" is I remember
kamailio[5009]: {2 1 INVITE [email protected]} ERROR: <script>: for 180 response the value of AVP "state_test_var" is I remember
kamailio[5011]: {2 1 INVITE [email protected]} ERROR: <script>: for 200 response the value of AVP "state_test_var" is I remember
kamailio[5010]: {1 1 ACK [email protected]} ERROR: <script>: for ACK the value of AVP "state_test_var" is <null>
kamailio[5004]: {1 2 BYE [email protected]} ERROR: <script>: for BYE the value of AVP "state_test_var" is <null>
kamailio[5007]: {2 2 BYE [email protected]} ERROR: <script>: for 200 response the value of AVP "state_test_var" is <null>
Here we can see for the INVITE, the 100 TRYING, 180 RINGING and 200 OK responses, state was maintained as the variable we set in the INVITE we were able to reference again.
(The subsequent BYE didn’t get state maintained because it’s not part of the transaction.)
I’m not a fan of Transcoding. It costs resources, often leads to reduced quality and adds latency.
Through some fancy SDP manipulating footwork we can often rejig the SDP order or limit the codecs we don’t support to cut down, or even remove entirely, the need for transcoding in the network.
There are no module parameters for SDP ops, we’ve just got to load the module with loadmodule “sdpops.so”
Use in Routing Logic
We’ll pickup where we left off on the Basic Stateless SIP Proxy use case (You can grab the basic code from that post), but this time we’ll remove PCMU (Aka G.711 μ-law) from the SDP body:
loadmodule "sdpops.so"
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
if(is_method("REGISTER")){
sl_reply("200", "Ok");
}
xlog("Received $rm to $ru - Forwarding");
append_hf("X-Proxied: You betcha\r\n");
#Remove PCMU (G.711 u-law) by it's SDP Payload ID
sdp_remove_codecs_by_id("0");
#Remove PCMU by name
sdp_remove_codecs_by_name("PCMU");
#Forard to new IP
forward("192.168.3.110");
}
onreply_route{
xlog("Got a reply $rs");
append_hf("X-Proxied: For the reply\r\n");
}
We can remove the codec either by it’s name (PCMU) or by it’s payload ID.
Before traversing the ProxyAfter traversing the proxy
For removing it by name we just specify the name:
#Remove PCMU by name
sdp_remove_codecs_by_name("PCMU");
And by payload ID:
#Remove PCMU (G.711 u-law) by it's SDP Payload ID
sdp_remove_codecs_by_id("0");
We may want to remove all but one codec, again super simple:
We’ve talked a bit in the Kamailio Bytes series about different modules we can use, but I thought it’d be useful to talk about setting up a SIP Proxy using Kamailio, and what’s involved in routing a message from Host A to Host B.
When we talk about proxying for the most part we’re talking about forwarding, let’s look at the steps involved:
Our Kamailio instance receives a SIP request (for simplicity we’ll assume an INVITE).
Kamailio looks at it’s routing logic to lookup where to forward the request to. You could find out where to send the request to from a lot of different sources, you could consult the Dialplan Module or Dispatcher Module, perform an SQL lookup, consult UsrLoc table to find a AoR, etc.
Add it’s own Via header (more on that later)
Forward the Request (aka Proxy it) to the destination selected
Let’s take a look at a very simple way we can do this with two lines in Kamailio to forward any requests to 192.168.1.110:
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
xlog("Received $rm to $ru - Forwarding");
#Forard to new IP
forward("192.168.1.110");
}
After we restart Kamailio and send a call (INVITE) to it let’s see how it handles it:
Let’s make a small modification, we’ll add a header called “X-Proxied” to the request before we forward it.
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
xlog("Received $rm to $ru - Forwarding");
append_hf("X-Proxied: You betcha\r\n");
#Forard to new IP
forward("192.168.1.110");
}
On the wire the packets still come from the requester, to the Proxy (Kamailio) before being forwarded to the forward destination (192.168.1.110):
We’ve now got a basic proxy that takes all requests to the proxy address and forwards it to an IP Address.
If you’re very perceptive you might have picked up the fact that the in-dialog responses, like the 100 Trying, the 180 Ringing and the 200 Ok also all went through the proxy, but if you look at syslog you’ll only see the initial request.
/usr/sbin/kamailio: Received INVITE to sip:[email protected]:5060 - Forwarding
So why didn’t we hit that xlog() route and generate a log entry for the replies?
But before we can talk too much about managing replies, let’s talk about Via…
It’s all about the Via
Before we can answer that question let’s take a look at Via headers.
The SIP Via header is added by a proxy when it forwards a SIP message onto another destination,
When a response is sent the reverse is done, each SIP proxy removes their details from the Via header and forwards to the next Via header along.
SIP Via headers in action
As we can see in the example above, each proxy adds it’s own address as a Via header, before it uses it’s internal logic to work out where to forward it to, and then forward on the INVITE.
Now because all our routing information is stored in Via headers when we need to route a Response back, each proxy doesn’t need to consult it’s internal logic to work out where to route to, but can instead just strip it’s own address out of the Via header, and then forward it to the next Via header IP Address down in the list.
Via headers are also used to detect looping, a proxy can check when it receives a SIP message if it’s own IP address is already in a Via header, if it is, there’s a loop there.
Managing Responses in Kamailio
By default Kamailio manages responses by looking at the Via header, if the top Via header is its own IP address, it strips it’s own Via header and forwards it onto the next destination in the Via header.
We can add our own logic into this by adding a new route called onreply_route{}
onreply_route{
xlog("Got a reply $rs");
append_hf("X-Proxied: For the reply\r\n");
}
Now we’ll create a log entry with the response code in syslog for each response we receive, and we’ll add a header on the replies too:
Recap
A simple proxy to forward INVITEs is easy to implement in Kamailio, the real tricky question is what’s the logic involved to make the decision,
Now we’ll put both together to create something functional you could use in your own deployments. (You’d often find it’s faster to use HTable to store and retrieve data like this, but that’s a conversation for another day)
The Project
We’ll build a SIP honeypot using Kamailio. It’ll listen on a Public IP address for SIP connections from people scanning the internet with malicious intent and log their IPs, so our real SIP softswitches know to ignore them.
We’ll use GeoIP2 to lookup the location of the IP and then store that data into a MySQL database.
Lastly we’ll create a routing block we can use on another Kamailio instance to verify if that the IP address of the received SIP message is not in our blacklist by searching the MySQL database for the source IP.
The Database
In this example I’m going to create a database called “blacklist” with one table called “baddies”, in MySQL I’ll run:
CREATE database blacklist;
CREATE TABLE `baddies` (
`id` INT NOT NULL AUTO_INCREMENT PRIMARY KEY,
`ip_address` INT unsigned UNIQUE,
`hits` INT,
`last_seen` DATETIME,
`ua` TEXT,
`country` TEXT,
`city` TEXT
);
I’ll setup a MySQL user to INSERT/UPDATE/SELECT data from the MySQL database.
For storing IP addresses in the database we’ll store them as unsigned integers, and then use the INET_ATON('127.0.0.1') MySQL command to encode them from dotted-decimal format, and the INET_NTOA('2130706433') to put them back into dotted decimal.
Modparams
Now we’ll need to configure Kamailio, I’ll continue on from where we left off in the last post on GeoIP2 as we’ll use that to put Geographic data about the IP before adding the MySQL and SQLOps modules:
# ----- SQL params -----
loadmodule "db_mysql.so"
loadmodule "sqlops.so"
#Create a new MySQL database connection called blacklist_db
modparam("sqlops","sqlcon","blacklist_db=>mysql://root:yourpassword@localhost/blacklist")
#Set timeouts for MySQL Connections
modparam("db_mysql", "ping_interval", 60)
modparam("db_mysql", "auto_reconnect", 1)
modparam("db_mysql", "timeout_interval", 2)
After loading db_mysql and sqlops we create a new object / connection called blacklist_db with our MySQL Database parameters.
Now after a restart we’ll be connected to our MySQL database.
Honeypot Routing Logic
Now we’ll create a route to log the traffic:
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
route(AddToBlacklist);
sl_reply('200', 'Sure thing boss!');
}
route[AddToBlacklist]{
xlog("Packet received from IP $si");
sql_xquery("blacklist_db", "insert into baddies (ip_address, hits, last_seen, ua, country, city) values (2130706433, 10, NOW(), 'testua2', 'Australia', 'Hobart');");
}
Now for each SIP message received a new record will be inserted into the database:
root@ip-172-31-8-156:/etc/kamailio# mysql -u root -p blacklist -e "select * from baddies;"
Enter password:
+----+------------+------+---------------------+---------+-----------+--------+
| id | ip_address | hits | last_seen | ua | country | city |
+----+------------+------+---------------------+---------+-----------+--------+
| 1 | 2130706433 | 10 | 2019-08-13 02:52:57 | testua2 | Australia | Hobart |
| 2 | 2130706433 | 10 | 2019-08-13 02:53:01 | testua2 | Australia | Hobart |
| 3 | 2130706433 | 10 | 2019-08-13 02:53:05 | testua2 | Australia | Hobart |
+----+------------+------+---------------------+---------+-----------+--------+
This is great but we’re not actually putting the call variables in here, and we’ve got a lot of duplicates, let’s modify our sql_xquery() to include the call variables:
Now we’re setting the IP Address value to the Source IP psedovariable ($si) and formatting it using the INET_ATON function in MySQL, setting the last_seen to the current timestamp and setting the user agent to the User Agent psedovariable ($ua).
Let’s restart Kamailio, truncate the data that’s currently in the DB, send some SIP traffic to it and then check the contents:
mysql -u root -p blacklist -e "select *, INET_NTOA(ip_address) from baddies;"
Here you can see we’re starting to get somewhere, the IP, UA and last_seen values are all now correct.
We’re getting multiple entries from the same IP though, instead we just want to increment the hits counter and set the last_seen to the current time, for that we’ll just update the SQL query to set the time to be NOW() and if that IP is already in the database to update the last_seen value and incriment the hits counter:
route[AddToBlacklist]{
xlog("Packet received from IP $si");
geoip2_match("$si", "src"))
sql_xquery("blacklist_db", "insert into baddies (ip_address, hits, last_seen, ua, country, city) values (INET_ATON('$si'), 1, NOW(), '$ua', '$gip2(src=>cc)', '$gip2(src=>city)') ON DUPLICATE KEY UPDATE last_seen = NOW(), hits = hits + 1;", "r_sql");
}
The only issue with this is if GeoIP2 doesn’t have a match, no record will be added in the database, so we’ll add a handler for that:
route[AddToBlacklist]{
xlog("Packet received from IP $si");
if(geoip2_match("$si", "src")){
sql_xquery("blacklist_db", "insert into baddies (ip_address, hits, last_seen, ua, country, city) values (INET_ATON('$si'), 1, NOW(), '$ua', '$gip2(src=>cc)', '$gip2(src=>city)') ON DUPLICATE KEY UPDATE last_seen = NOW(), hits = hits + 1;", "r_sql");
}else{ ##If no match in GeoIP2 leave Country & City fields blank
sql_xquery("blacklist_db", "insert into baddies (ip_address, hits, last_seen, ua, country, city) values (INET_ATON('$si'), 1, NOW(), '$ua', '', '') ON DUPLICATE KEY UPDATE last_seen = NOW(), hits = hits + 1;", "r_sql");
}
}
Now let’s check our database again and see how the data looks:
mysql -u root -p blacklist -e "select *, INET_NTOA(ip_address) from baddies;"
Perfect! Now we’re inserting data into our blacklist from our honeypot. Now we’ll configure a new routing block we can use on another Kamailio instance to see if an IP is in the blacklist.
I left this running on my AWS box for a few hours, and lots of dodgy UAs dropped in to say hello, one of which was very insistent on making calls to Poland…
Querying the Data
Now we’ve got a blacklist it’s only useful if we block the traffic from our malicous actors who we’ve profiled in the database.
You could feed this into BGP to null route the traffic, or hook this into your firewall’s API, but we’re going to do this in Kamailio, so we’ll create a new routing block we can use on a different Kamailio instance – Like a production one – to see if the IP it just received traffic from is in the blacklist.
We’ve already spoken about querying databases in the SQLops Kamailio bytes, but this routing block will query the blacklist database, and if the sender is in the database, one or more records will be returned, so we know they’re bad and will drop their traffic:
route[CheckBlacklist]{
xlog("Checking blacklist for ip $si");
#Define a variable containing the SQL query we'll run
$var(sql) = "select INET_NTOA(ip_address) as ip_address from baddies;";
#Log the SQL query we're going to run to syslog for easy debugging
xlog("Query to run is $var(sql)");
#Query blacklist_db running the query stored in $var(sql) and store the result of the query to result_sql
sql_query("blacklist_db", "$var(sql)", "result_sql");
#If more than 0 records were returned from the database, drop the traffic
if($dbr(result_sql=>rows)>0){
xlog("This guy is bad news. Dropping traffic from $si");
exit;
}else{
xlog("No criminal record for $si - Allowing to progress");
}
}
This Honeypot use case just put those elements together.
In reality a far better implementation of this would use HTable to store this data, but hopefully this gives you a better understanding of how to actually work with data.
Final Note
I wrote this post about a week ago, and left the config running on an AWS box. I was getting hits to it within the hour, and in the past week I’ve had 172 IPs come and say hello, and some like the FriendlyScanner instance at 159.65.220.215 has sent over 93,000 requests:
We’ve touched a tiny bit on basic database functionality in Kamailio, using MySQL to store User Data for authentication, ACLs or Dispatcher entries.
But with all those we were using Databases to load the config / dynamic data for a module.
We’ll build upon that, to connect to a Database that we can INSERT, UPDATE and SELECT data from within the dialplan.
For today’s example we’ll lookup the To address from a SIP INVITE and send back
Heads Up
There’s a lot of different use cases for reading and writing data from a database, but Kamailio also has a lot of native modules that handle this better, for example:
You might want to store a record of each INVITE and BYE you recieve for accounting, a better option is to use the Accounting Module in Kamailio.
You might want to authenticate user’s based on ACLs stored in a database, a better option would be to use Permissions Module.
User authentication info is best handled by Auth DB module.
The Dialplan module handles number translation really well and is lightning quick.
Just keep this in mind before jumping in that a lot of use cases have already been covered by a Kamailio module.
The Architecture
For today’s example we’ll be using MySQL as the database backend (db_mysl), but the db_mysql module simply connects us to a database, a bit like ODBC.
The real MVP is the SQLops module, that does all the heavy lifting by running the queries and managing the responses.
The majority of this config would work fine for other database backends, like PostGres, MongoDB, Oracle, etc.
I’ll demonstrate this same setup using different database backends in future posts.
MySQL Database
Before we get too excited we’ll need to setup a database to query. I’ll create a database called dyamic_routing with a table called routing storing source / destinations.
CREATE DATABASE phonebook;
USE phonebook;
CREATE TABLE contacts (
id INT NOT NULL AUTO_INCREMENT PRIMARY KEY,
source TEXT,
name TEXT
);
INSERT INTO contacts VALUES (1, '200', 'Nick Deskphone');
I’ll setup a MySQL user to INSERT/UPDATE/SELECT data from the MySQL database the normal way.
Modparam
The module parameters for connecting to a database backend are fairly straight forward, but we’ll go into a bit of depth here to drive home the point.
# ----- SQL params -----
loadmodule "db_mysql.so"
loadmodule "sqlops.so"
#Create a new MySQL database connection called contacts_db
modparam("sqlops","sqlcon","contacts_db=>mysql://root:youshouldntrealluseroot@localhost/phonebook")
#Set timeouts for MySQL Connections
modparam("db_mysql", "ping_interval", 60)
modparam("db_mysql", "auto_reconnect", 1)
modparam("db_mysql", "timeout_interval", 2)
First off we load the two modules we need for this, db_mysql and sqlops. This is fairly self explanatory, if we were using db_postgres, db_mongodb or even db_text we’d load them instead of db_mysql.
The sqlops “sqlcon” modparam is where we define our MySQL database connection.
In this case we create a new database connection object called contacts_db– We can have connections to multiple databases, hence requiring a name.
The MySQL URL is fairly straightforward, database type, username, password, host and database:
mysql://root:password@localhost/phonebook
In production obviously you shouldn’t use root as the user account to log into the database, and lock it down to only the source IP of your Kamailio instance and with only the permissions it needs. (If it’s just selecting data there’s no need for GRANT…)
Basic Query
Now we’ve created a database connection it’s time to start using it.
request_route {
if(method=="INVITE"){
xlog("New request to $tU");
#Query database object called "contacts_db", run the below query and store the output to a variable called result_sql
sql_query("contacts_db", "select * from contacts;", "result_sql");
#output number of rows in database returned
xlog("number of rows in table is $dbr(result_sql=>rows)\n");
}
}
If the method is an INVITE we’ll query the database object called “contacts_db” to run the query select * from contacts;
We’ll then output the number of rows in the table to xlog.
The query actually happens in the sql_query() command, which takes the name of the database object ( contacts_db ), the query itself ( select * from contacts; ) and stores it into a variable called result_sql.
Finally xlog references the variable we stored our result in (result_sql) using the $dbr() handler to output the number of rows in the table.
If you save this and send an INVITE to any destination and watch the syslog you should see something along the lines of this:
/usr/sbin/kamailio[7815]: ERROR: : New request to 200
/usr/sbin/kamailio[7815]: ERROR: <script>: number of rows in table is 1
This means we’ve got a connection to the database and we can run queries.
Accessing the Output
Now we’ve got the data back from the database and stored it in result_sql we probably want to do something with the outputted data.
By wrapping the result_sql variable in the $dbr() tags we can access it’s jucy insides, let’s take a look:
#output number of columns
xlog("Result has $dbr(result_sql=>cols) Columns");
#output number of rows
xlog("Result has $dbr(result_sql=>rows) rows");
#output contents of row 0, column 2
xlog("Contents of row 0 col 2 is $dbr(result_sql=>[0,2]) ");
#output name of column 2
xlog("name of column 2 is $dbr(result_sql=>colname[2]) ");
If we add this after our last xlog line, restart Kamailio and view syslog it should look something like this:
/usr/sbin/kamailio[8249]: ERROR: <script>: New request to 200
/usr/sbin/kamailio[8249]: ERROR: <script>: number of rows in table is 1
/usr/sbin/kamailio[8249]: ERROR: <script>: Result has 3 Columns
/usr/sbin/kamailio[8249]: ERROR: <script>: Result has 1 rows
/usr/sbin/kamailio[8249]: ERROR: <script>: Contents of row 0 column 2 is Nick Deskphone
Now we can see the data in the result we’ll start to refine this down a bit, we’ll begin by limiting the SQL query to search for the called number.
For this we’ll update the sql_query(); function to:
sql_query("contacts_db", "select * from contacts where source = $tU;", "result_sql");
This will include the the To URI Username pseudo variable in our query, so will only return results if the number we dial has one or more matching “source” entries in the database.
If we dial 200 the query that’s actually run against the database will look like this:
select * from contacts where source = '200';
Now once we save and try again our traffic will look the same, except it’ll only return data if we dial 200, if we dial 201 the SQL server won’t have any matching results to return:
/usr/sbin/kamailio[9069]: ERROR: : New request from 2029
/usr/sbin/kamailio[9069]: ERROR: number of rows in table is 0
/usr/sbin/kamailio[9069]: ERROR: Result has 0 Columns
So that’s all well and good but we haven’t really got the data easily yet, while we’re outputting the contents of row 0 col 2 to syslog, it’s not going to handle multiple results being returned, or 0 results being returned, so we need a better way to handle this.
We’ll use a for loop to loop through all the results returned and output the second column of each (the “name” field in the database).
#Loop through results
#Create variable i to use as the counter
$var(i) = 0;
#While the contents of row i, position 2, is not null:
while ($dbr(result_sql=>[$var(i),2]) != $null) {
#Output row i, position 2 (name)
xlog("name is $dbr(result_sql=>[$var(i),2])");
#increment i by 1
$var(i) = $var(i) + 1;
}
So while the contents of row i, position 2, is not null, we’ll output the contents and increment i to get the next row in the database until there are none left.
Now we can give our code a bit of a clean up:
request_route {
if(method=="INVITE"){
xlog("New request from $tU");
#Query database object called "contacts_db", run the below query and store the output to a variable called result_sql
sql_query("contacts_db", "select * from contacts where source = $tU;", "result_sql");
#Loop through results
#Create variable i to use as the counter
$var(i) = 0;
#While the contents of row i, position 2, is not null:
while ($dbr(result_sql=>[$var(i),2]) != $null) {
#Output row i, position 2 (name)
xlog("name $dbr(result_sql=>[$var(i),2])");
#increment i by 1
$var(i) = $var(i) + 1;
}
}
if(method=="REGISTER"){ sl_reply('200', 'OK'); }
}
I’ve removed many of our xlog entries we put in for debugging and also added a handler to handle REGISTER requests to keep my IP phone happy.
Now if we make a call to number 200:
/usr/sbin/kamailio[9686]: ERROR: New request from 200
/usr/sbin/kamailio[9686]: ERROR: name Nick Deskphone
And for comparison a call to 201 (no matching database entry):
/usr/sbin/kamailio[9686]: ERROR: New request from 200
Using the Resulting Output
Now we’ve got access to the data from the database let’s do something with it.
Inside our loop we’ll send a reply to the SIP requester, with a 410 “Gone” response with the body containing the data returned from the database:
#Loop through results
#Create variable i to use as the counter
$var(i) = 0;
#While the contents of row i, position 2, is not null:
while ($dbr(result_sql=>[$var(i),2]) != $null) {
#Output row i, position 2 (name)
xlog("name $dbr(result_sql=>[$var(i),2])");
$var(name) = $dbr(result_sql=>[$var(i),2]);
#increment i by 1
$var(i) = $var(i) + 1;
#Reply with a 410 (User Gone) response with the name returned from the database
sl_reply("410", "Sorry $var(name) has gone home");
exit;
}
Now calls to 200 will get the message “Sorry Nick desk phone has gone home”.
Lastly we probably want to only loop through the output if there’s more than one row returned from the database, so we’ll put the looping code in an if statement that evaluates if the number of returned rows from the database is 1 or more, and if not just send a 404 response:
#if one or more results are returned from the database
if($dbr(result_sql=>rows)>0){
#Loop through results
#Create variable i to use as the counter
$var(i) = 0;
#While the contents of row i, position 2, is not null:
while ($dbr(result_sql=>[$var(i),2]) != $null) {
#Output row i, position 2 (name)
xlog("name $dbr(result_sql=>[$var(i),2])");
$var(name) = $dbr(result_sql=>[$var(i),2]);
#increment i by 1
$var(i) = $var(i) + 1;
#Reply with a 410 (User Gone) response with the name returned from the database
sl_reply("410", "Sorry $var(name) has gone home");
exit;
}
}else{
#if 0 results are returned from database
sl_reply("404", "Never heard of them");
}
INSERT, DELETE, UPDATE, etc
Although we only covered SELECT, queries that don’t return data like an INSERT, UPDATE, DELETE etc, can all be run the same way but we just don’t need to worry about managing the returned data.
For example we could delete a record using:
sql_query("contacts_db", "delete * from contacts where source = $tU;");
We don’t even need to store the output unless we need to.
Summary
Hopefully you’ve now got an idea how to query data from a database and view / manipulated the returned data.
Generally Kamailio functions as a SIP router, receiving SIP messages and then responding with SIP.
Sometimes we may have a use case where we need to interact with Kamailio but with a request that isn’t a SIP message.
You’ve got a motion activated IP Camera, and you want to send an alert to a SIP phone if it detects motion.
The problem? The IP camera doesn’t speak SIP. but it can send an HTTP request if it detects motion.
Enter xHTTP!
We’ll get Kamailio to listen for HTTP requests and send an instant message using method “MESSAGE” to a SIP phone to let someone know there’s been motion.
Use Case
The sending of the message is fairly straight forward, we’ll use the UAC module to perform a uac_req_send() and put it in it’s own routing block called “SEND_MESSAGE”, which we’ll add after the request_route{} block:
Now when we call the route ROUTE(SEND_MESSAGE); the SIP phone at 10.0.1.5 will get a message that pops up on the screen.
So the next step is getting something other than a SIP request to call our SEND_MESSAGE route.
For this we’ll use the xHTTP module, a barebones HTTP server for Kamailio.
It’s requirements are pretty easy, sl_reply needs to be loaded before xHTTP, and you may need to add tcp_accept_no_cl=yes to your core settings (aka Global Parameters at the top).
The two lines we’ll need to load and configure the module are equally as easy:
The url_match modparam just means that any HTTP request has to start with /sip/ in the URL.
Finally we’ll define an event route to handle any xHTTP requests after our request_route{} block:
event_route[xhttp:request] {
xlog("Recieved HTTP request with request $hu"); #Write to log the URL of http request.
xhttp_reply("200", "OK", "text/html", "<html><body>OK</body></html>"); #Send HTTP response
}
Now if we restart Kamailio, and open a browser pointed to the IP of your Kamailio server, port 5060 /sip/Hello World you should get a response of “OK” in the browser:
Perfect, we now have an HTTP server in Kamailio, and we can read the HTTP request URL into a variable.
Next up we can call the route(SEND_MESSAGE) and our SIP phone will get a message
event_route[xhttp:request] {
xlog("Recieved HTTP request with request $hu"); #Write to log the URL of http request.
xhttp_reply("200", "OK", "text/html", "<html><body>OK</body></html>"); #Send HTTP response
route(SEND_MESSAGE);
}
Presto! When we call that URL (http://your-kamailio-ip:5060/sip/whatever) a SIP SIMPLE MESSAGE is sent.
But why stop there, let’s make this a bit prettier, we’ll set the message to equal the part of the HTTP request after the /sip/ so we can send custom data, replace the %20 with underscores and send it:
route[SEND_MESSAGE]{
$uac_req(method)="MESSAGE";
$uac_req(ruri)="sip:192.168.3.227:5060";
$uac_req(furi)="sip:nickvsnetworking.com";
$uac_req(turi)="sip:thisphone";
$uac_req(callid)=$(mb{s.md5});
$uac_req(hdrs)="Subject: Test\r\n";
$uac_req(hdrs)=$uac_req(hdrs) + "Content-Type: text/plain\r\n";
$uac_req(body)=$var(message);
uac_req_send();
}
event_route[xhttp:request] {
xlog("Recieved HTTP request with request $hu"); #Write to log the URL of http request.
$var(message) = $hu; #Set variable $var(message) to equal the URL of http request.
$var(message) = $(var(message){s.substr,5,0}); #Cut off first 5 chars to exclude the /sip/ prefix from the HTTP request
$var(message) = $(var(message){s.replace,%20,_}); #Replace %20 of space in HTTP request with Underscore _ for spaces
xlog("var message is $var(message)"); #Write to log the http request minus the /sip/ prefix
xhttp_reply("200", "OK", "text/html", "<html><body>OK</body></html>"); #Send HTTP response
route(SEND_MESSAGE); #Call the SEND_OPTIONS route (See above)
}
We’ll also set our core settings / gloabal parameters to listen on TCP port 80 as well as UDP port 5060 so we don’t need to specify the port in the browser:
Hopefully by now you can see some of the cool things you can do with the HTTP module. Kamailio is so much more than just a SIP router / proxy, and these external modules being able to interface with it give you so many options.
Want to offer webhooks to customers to control their calls? xHTTP can do that!
Want to play a message to all users on calls announcing to them lunch is ready? RTPengine and xHTTP can do that too!
If you’re planning on using this in production you probably want to automate the pulling of this data on a regular basis and keep it in a different directory.
I’ve made a very simple example Kamailio config that shows off some of the features of GeoIP2’s logic and what can be shown, so let’s look at the basics of the module:
if(geoip2_match("$si", "src")){
xlog("Packet received from IP $si");
xlog("Country is: $gip2(src=>cc)\n");
}
If we put this at the top of our request_route block every time we recieve a new request we can see the country from which the packet came from.
Let’s take a look at the output of syslog (with my IP removed):
#> tail -f /var/log/syslog
ERROR: <script>: Packet received from IP 203.###.###.###
ERROR: <script>: Country is: AU
ERROR: <script>: City is: Melbourne
ERROR: <script>: ZIP is: 3004
ERROR: <script>: Regc is: VIC
ERROR: <script>: Regn is: Victoria
ERROR: <script>: Metro Code is: <null>
We can add a bunch more smarts to this and get back a bunch more variables, including city, ZIP code, Lat & Long (Approx), timezone, etc.
if(geoip2_match("$si", "src")){
xlog("Packet received from IP $si");
xlog("Country is: $gip2(src=>cc)\n");
xlog("City is: $gip2(src=>city)");
xlog("ZIP is: $gip2(src=>zip)");
xlog("Regc is: $gip2(src=>regc)");
xlog("Regn is: $gip2(src=>regn)");
xlog("Metro Code is: $gip2(src=>metro)");
if($gip2(src=>cc)=="AU"){
xlog("Traffic is from Australia");
}
}else{
xlog("No GeoIP Match for $si");
}
#> tail -f /var/log/syslog
ERROR: <script>: Packet received from IP ###.###.###.###
ERROR: <script>: Country is: AU
ERROR: <script>: City is: Melbourne
ERROR: <script>: ZIP is: 3004
ERROR: <script>: Regc is: VIC
ERROR: <script>: Regn is: Victoria
ERROR: <script>: Metro Code is: <null>
Using GeoIP2 you could use different rate limits for domestic users vs overseas users, guess the dialling rules based on the location of the caller and generate alerts if accounts are used outside their standard areas.
We’ll touch upon this again in our next post on RTPengine where we’ll use an RTPengine closes to the area in which the traffic originates.
I’m a big fan of RTPengine, and I’ve written a bit about it in the past.
Let’s say we’re building an Australia wide VoIP network. It’s a big country with a lot of nothing in the middle. We’ve got a POP in each of Australia’s capital cities, and two core softswitch clusters, one in Melbourne and one in Sydney.
These two cores will work fine, but a call from a customer in Perth, WA to another customer in Perth, WA would mean their RTP stream will need to go across your inter-caps to Sydney or Melbourne only to route back to Perth.
That’s 3,500Km each way, which is going to lead to higher latency, wasted bandwidth and decreased customer experience.
What if we could have an RTPengine instance in our Perth POP, handling RTP proxying for our Perth customers? Another in Brisbane, Canberra etc, all while keeping our complex expensive core signalling in just the two locations?
RTPengine to the rescue!
Preparing our RTPEngine Instances
In each of our POPs we’ll spin up a box with RTPengine,
The only thing we’d do differently is set the listen-ng value to be 0.0.0.0:2223 and the interface to be the IP of the box.
By setting the listen-ng value to 0.0.0.0:2223 it’ll mean that RTPengine’s management port will be bound to any IP, so we can remotely manage it via it’s ng-control protocol, using the rtpengine Kamailio module.
Naturally you’d limit access to port 2223 only to allowed devices inside your network.
Next we’ll need to add the details of each of our RTP engine instances to MySQL, I’ve used a different setid for each of the RTPengines. I’ve chosen to use the first digit of the Zipcode for that state (WA’s Zipcodes / Postcodes are in the format 6xxx while NSW postcodes are look like 2xxx), we’ll use this later when we select which RTPengine instances to use.
I’ve also added localhost with setid of 0, we’ll use this as our fallback route if it’s not coming from Australia.
Bingo, we’re connected to three RTPengine instances,
Next up we’ll use the Geoip2 module to determine the source of the traffic and route to the correct, I’ve touched upon the Geoip2 module’s basic usage in the past, so if you’re not already familiar with it, read up on it’s usage and we’ll build upon that.
We’ll load GeoIP2 and run some checks in the initial request_route{} block to select the correct RTPengine instance:
if(geoip2_match("$si", "src")){
if($gip2(src=>cc)=="AU"){
$var(zip) = $gip2(src=>zip);
$avp(setid) = $(var(zip){s.substr,0,1});
xlog("rtpengine setID is $avp(setid)");
}else{
xlog("GeoIP not in Australia - Using default RTPengine instance");
set_rtpengine_set("0");
}
}else{
xlog("No GeoIP Match - Using default RTPengine instance");
set_rtpengine_set("0");
}
In the above example if we have a match on source, and the Country code is Australia, the first digit of the ZIP / Postcode is extracted and assigned to the AVP “setid” so RTPengine knows which set ID to use.
In practice an INVITE from an IP in WA returns setID 6, and uses our RTPengine in WA, while one from NSW returns 2 and uses one in NSW. In production we’d need to setup rules for all the other states / territories, and generally have more than one RTPengine instance in each location (we can have multiple instances with the same setid).
Hopefully you’re starting to get an idea of the fun and relatively painless things you can achieve with RTPengine and Kamailio!
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