RTPengine has an API / control protocol, which is what Kamailio / OpenSER uses to interact with RTPengine, called the ng Control Protocol.
Connection is based on Bencode encoded data and communicates via a UDP socket.
I wrote a simple Python script to pull active calls from RTPengine, code below:
#Quick Python library for interfacing with Sipwise's fantastic rtpengine - https://github.com/sipwise/rtpengine
#Bencode library from https://pypi.org/project/bencode.py/ (Had to download files from webpage (PIP was out of date))
import bencode
import socket
import sys
import random
import string
sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
server_address = ('188.0.169.13', 2224) #Your server address
cookie = "0_2393_6"
data = bencode.encode({'command': 'list'})
message = str(cookie) + " " + str(data)
print(message)
sent = sock.sendto(message, server_address)
print('waiting to receive')
data, server = sock.recvfrom(4096)
print('received "%s"' % data)
data = data.split(" ", 1) #Only split on first space
print("Cookie is: " + str(data[0]))
print("Data is: " + str(bencode.decode(data[1])))
print("There are " + str(len(bencode.decode(data[1])['calls'])) + " calls up on RTPengine at " + str(server_address[0]))
for calls in bencode.decode(data[1])['calls']:
print(calls)
cookie = "1_2393_6"
data = bencode.encode({'command': 'query', 'call-id': str(calls)})
message = str(cookie).encode('utf-8') + " ".encode('utf-8') + str(data).encode('utf-8')
sent = sock.sendto(message, server_address)
print('\n\nwaiting to receive')
data, server = sock.recvfrom(8192)
data = data.split(" ", 1) #Only split on first space
bencoded_data = bencode.decode(data[1])
for keys in bencoded_data:
print(keys)
print("\t" + str(bencoded_data[keys]))
sock.close()
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 = $tU;");
We don’t even need to store the output unless we need to.
Summary
Hopefully you’ve now got an idea how to query data from a database and view / manipulated the returned data.
If you’re planning on using this in production you probably want to automate the pulling of this data on a regular basis and keep it in a different directory.
I’ve made a very simple example Kamailio config that shows off some of the features of GeoIP2’s logic and what can be shown, so let’s look at the basics of the module:
if(geoip2_match("$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!
Siremis is a web interface for Kamailio, created by the team at Asipto, who contribute huge amounts to the Kamailio project.
Siremis won’t create your Kamailio configuration file for you, but allows you to easily drive the dynamic functions like dialplan, subscribers, dispatcher, permissions, etc, from a web interface.
Siremis essentially interfaces with the Kamailio database, kamcmd and kamctl to look after your running Kamailio instance.
Installation
I’ll be installing on Ubuntu 18.04, but for most major distributions the process will be the same. We’re using PHP7 and Apache2, which are pretty much universal available on other distros.
First we need to install all the packages we require:
Next we’ll download Siremis from the Git repo, and put it into a folder, which I’ve named the same as my Kamailio version.
cd /var/www/html/ git clone https://github.com/asipto/siremis kamailio-5.1.2
Now we’ll move into the directory we’ve created (called kamailio-5.1.2) and build the apache2 config needed:
cd kamailio-5.1.2/ make apache24-conf
This then gives us a config except we can put into our Apache virtual host config file:
We can now copy and paste this into the end of an existing or new Apache virtual host file.
If this is a fresh install you can just pipe the output of this into the config file directly:
make apache24-conf >> /etc/apache2/sites-available/000-default.conf service apache2 restart
Now if you browse to http://yourserverip/siremis you should be redirected to http://yourserverip/siremis/install and have a few errors, that’s OK, it means our Apache config is working.
Next we’ll set the permissions, create the folders and .htaccess. The Siremis team have also created make files to take care of this too, so we can just run them to set everything up:
make prepare24 make chown
With that done we can try browsing to our server again ( http://yourserverip/siremis ) and you should hit the installation wizard:
Now we’ll need to setup our database, so we can read and write from it.
We’ll create new MySQL users for Kamailio and Seremis:
mysql> GRANT ALL PRIVILEGES ON siremis.* TO siremis@localhost IDENTIFIED BY 'siremisrw';
mysql> CREATE USER 'kamailio'@'localhost' IDENTIFIED BY 'my5yhtY7zPJzV8vu';
mysql> GRANT ALL PRIVILEGES ON * . * TO 'kamailio'@'localhost';
mysql> FLUSH PRIVILEGES;
Next up we’ll need to configure kamctlrc so it knows the database details, we covered this the Security in Practice tutorial.
We’ll edit /etc/kamalio/kamctlrc and add our database information:
Once that’s done we can create the database and tables using kamdbctl the database tool:
kamdbctl create
I’ve selected to install the optional tables for completeness.
Once this is done we can go back to the web page and complete the installation wizard:
We’ll need to fill the password for the Siremis DB we created and for the Kamailio DB, and ensure all the boxes are ticked.
Next, Next, Next your way through until you hit the login page, login with admin/admin and you’re away!
Troubleshooting
If you have issues during the installation you can re-run the installation web wizard by removing the install.lock file in /var/www/html/kamailio-5.1.2/siremis
You can also try dropping the Siremis database and getting the installer to create it again for you:
HTable is Kamailio’s implimentation of Hash Tables a database-like data structure that runs in memory and is very quick.
It’s uses only become apparent when you’ve become exposed to it.
Let’s take an example of protecting against multiple failed registration attempts.
We could create a SQL database called registration attempts, and each time one failed log the time and attempted username.
Then we could set it so before we respond to traffic we query the database, find out how many rows there are that match the username being attempted and if it’s more than a threshold we set we send back a rate limit response.
The problem is that’s fairly resource intensive, the SQL data is read and written from disks and is slow to do both.
Enter HTable, which achieves the same thing with an in-memory database, that’s lightning fast.
Basic Setup
We’ll need to load htable and create an htable called Table1 to store data in:
$sht(MessageCount=>test) is the logical link to the Htable called MessageCount with a key named test. We’re making that equal itself + 1.
We’re then outputting the content of $sht(MessageCount=>test) to xlog too so we can see it’s value in Syslog.
Now each time a new dialog is started the MessageCount htable key “test” will be incremented.
We can confirm this in Syslog:
ERROR: : MessageCount is 1 ERROR: : MessageCount is 2
We can also check this in kamcmd too:
htable.dump MessageCount
Here we can see in MessageCount there is one key named “test” with a value of 6, and it’s an integer. (You can also store Strings in HTable).
So that’s all well and pointless, but let’s do make it a bit more useful, report on how many SIP transactions we get per IP. Instead of storing our values with the name key “test” we’ll name it based on the Source IP of the message, which lives in Psedovariable $si for Source IP Address.
I’m calling the boilerplate AUTH block, and I’ve added some logic to increment the AuthCount for each failed auth attempt, and reset it to $null if authentication is successful, thus resetting the counter for that IP Address.
Now we’ve done that we need to actually stop the traffic if it’s failed too many times. I’ve added the below check into REQINIT block, which I call at the start of processing:
if($sht(AuthCount=>$si) > 5){
xlog("$si is back again, rate limiting them...");
sl_send_reply("429", "Rate limiting");
exit;
}
Now if AuthCount is more than 5, it’ll respond with a Rate Limiting response.
Because in our modparam() setup for AuthCount we set an expiry, after 360 seconds (10 minutes), after 10 minutes all will be forgiven and our blocked UA can register again.
Advanced Usage / Notes
So now we’ve got Kamailio doing rate limiting, it’s probably worth mentioning the Pike module, which can also be used.
You’ll notice if you reboot Kamailio all the htable values are lost, that’s because the hashes are stored in memory, so aren’t persistent.
You have a few options for making this data persistent,
By using DMQ you can Sync data between Kamailio instances including htable values.
kamcmd can view, modify & manipulate htable values.
As we’ve seen before we can dump the contents of an htable using:
kamcmd htable.dump MessageCount
We can also add new entries & modify existing ones:
kamcmd htable.seti MessageCount ExampleAdd s:999
htable.seti is for setting integer values, we can also use htable.sets to set string values:
htable.sets MessageCount ExampleAdd Iamastring
We can also delete values from here too, which can be super useful for unblocking destinations manually:
htable.delete MessageCount ExampleAdd
As always code from this example is on GitHub. (Please don’t use it in production without modification, Authentication is only called on Register, and it’s just built upon the previous tutorials).
There are a number of ways to feed Homer data, in this case we’re going to use Kamailio, which has a HEP module, so when we feed Kamailio SIP data it’ll use the HEP module to encapsulate it and send it to the database for parsing on the WebUI.
We won’t actually do any SIP routing with Kamailio, we’ll just use it to parse copies of SIP messages sent to it, encapsulate them into HEP and send them to the DB.
We’ll be doing this on the same box that we’re running the HomerUI on, if we weren’t we’d need to adjust the database parameters in Kamailio so it pushes the data to the correct MySQL database.
Next we’ll need to configure captagent to capture data and feed it to Kamailio. There’s two things we’ll need to change from the default, the first is the interface we capture on (By default it’s eth0, but Ubuntu uses eth33 as the first network interface ID) and the second is the HEP destination we send our data to (By default it’s on 9061 but our Kamailio instance is listening on 9060).
We’ll start by editing captagent’s socket_pcap.xml file to change the interface we capture on:
vi /etc/captagent/socket_pcap.xml
Next we’ll edit the port that we send HEP data on
vi /etc/captagent/transport_hep.xml
And finally we’ll restart captagent
/etc/init.d/captagent
Now if we send SIP traffic to this box it’ll be fed into HOMER.
In most use cases you’d use a port mirror so you may need to define the network interface that’s the destination of the port mirror in socket_pcap.xml
HOMER is a popular open source SIP / RTP debug / recording tool.
It’s architecture is pretty straight forward, we have a series of Capture Agents feeding data into a central HOMER Capture Server, which runs a database (today we’re using MySQL), a Homer-UI (Running on Apache), a Homer-API (Also running on Apache) and a HEP processor, which takes the HEP encoded data from the Capture Agents and runs on Kamailio. (That’s right, I’m back rambling about Kamailio)
So this will get the web interface and DB backend of HOMER setup,
For HOMER to actually work you’ll need to feed it data, in the next tutorial we’ll cover configuring a capture agent to feed the HEP processor (Kamailio) which we’ll also setup, but for now we’ll just setup the web user interface for HOMER, API and Database.
Now we’ll restart Kamailio and use kamcmd to check the status of our rtpengine instance:
kamcmd rtpengine.show all
All going well you’ll see something like this showing your instance:
Putting it into Practice
If you’ve ever had experience with the other RTP proxies out there you’ll know you’ve had to offer, rewrite SDP and accept the streams in Kamailio.
Luckily rtpengine makes this a bit easier, we need to call rtpengine_manage(); when the initial INVITE is sent and when a response is received with SDP (Like a 200 OK).
So for calling on the INVITE I’ve done it in the route[relay] route which I’m using:
And for the reply I’ve simply put a conditional in the onreply_route[MANAGE_REPLY] for if it has SDP:
Kamailio’s permissions module is simple to use, and we’ve already touched upon it in the security section in our Kamailio 101 series, but I thought I’d go over some of it’s features in more detail.
At it’s core, Kamailio’s Permissions module is a series of Access Control Lists (ACLs) that can be applied to different sections of your config.
We can manage permissions to do with call routing, for example, is that source allowed to route to that destination.
We can manage registration permissions, for example, is this subnet allowed to register this username.
We can manage URI permissions & address permissions to check if a specific SIP URI or source address is allowed to do something.
We’ll touch on a simple IP Address based ACL setup in this post, but you can find more information in the module documentation itself.
The Setup
We’ll be using a database backend for this (MySQL), setup the usual way.
We’ll need to load the permissions module and setup it’s basic parameters, for more info on setting up the database side of things have a look here.
Next we’ll need to add some IPs, we could use Serimis for this, or a straight MySQL INSERT, but we’ll use kamctl to add them. (kamcmd can reload addresses but doesn’t currently have the functionality to add them)
The above example we added a two new address entries,
The first one added a new entry in group 250 of “10.8.203.139”, with a /32 subnet mask (Single IP), on port 5060 with the label “TestServer”,
The second one we added to group 200 was a subnet of 192.168.1.0 with a /24 subnet mask (255 IPs), on port 5060 with the label “OfficeSubnet”
On startup, or when we manually reload the addressTable, Kamailio grabs all the records and stores them in RAM. This makes lookup super fast, but the tradeoff is you have to load the entries, so changes aren’t immediate.
Let’s use Kamcmd to reload the entries and check their status.
kamcmd permissions.addressReload
kamcmd permissions.addressDump
kamcmd permissions.subnetDump
You should see the single IP in the output of the permissions.addressDump and see the subnet on the subnetDump:
Usage
It’s usage is pretty simple, combined with a simple nested if statement.
if (allow_source_address("200")) {
xlog("Coming from address group 200");
};
if (allow_source_address("250")) {
xlog("Coming from address group 250");
};
The above example just outputs to xlog with the address group, but we can expand upon this to give us our ACL service.
if (allow_source_address("200")) {
xlog("Coming from address group 200");
}else if (allow_source_address("250")) {
xlog("Coming from address group 250");
}else{
sl_reply("401", "Address not authorised");
exit;
}
If we put this at the top of our Kamailio config we’ll reply with a 401 response to any traffic not in address group 200 or 250.
Kamalio’s dialplan is a bit of a misleading title, as it can do so much more than just act as a dialplan.
At it’s core, it runs transformations. You feed it a value, if the value matches the regex Kamailio has it can either apply a transformation to that value or return a different value.
Adding to Config
For now we’ll just load the dialplan module and point it at our DBURL variable:
loadmodule "dialplan.so"
modparam("dialplan", "db_url", DBURL); #Dialplan database from DBURL variable
Restart Kamailio and we can get started.
Basics
Let’s say we want to take StringA and translate it in the dialplan module to StringB, so we’d add an entry to the database in the dialplan table, to take StringA and replace it with StringB.
Now we’ll fire up Kamailio, open kamcmd and reload the dialplan, and dump out the entries in Dialplan ID 1:
dialplan.reload dialplan.dump 1
You should see the output of what we just put into the database reflected in kamcmd:
Now we can test our dialplan translations, using Kamcmd again.
dialplan.translate 1 StringA
All going well Kamailio will match StringA and return StringB:
So we can see when we feed in String A, to dialplan ID 1, we get String B returned.
Database Structure
There’s a few fields in the database we populated, let’s talk about what each one does.
dpid
dpid = Dialplan ID. This means we can have multiple dialplans, each with a unique dialplan ID. When testing we’ll always need to specific the dialplan ID we’re using to make sure we’re testing with the right rules.
priority
Priorities in the dialplan allow us to have different weighted priorities. For example we might want a match all wildcard entry, but more specific entries with lower values. We don’t want to match our wildcard failover entry if there’s a more specific match, so we use priorities to run through the list, first we try and match the group with the lowest number, then the next lowest and so on, until a match is found.
match_op
match_op = Match Operation. There are 3 options:
0 – string comparison;
1 – regular expression matching (pcre);
2 – fnmatch (shell-like pattern) matching
In our first example we had match_op set to 0, so we exactly matched “StringA”. The real power comes from Regex Matching, which we’ll cover soon.
match_exp
match_exp = Match expression. When match_op is set to 0 this matches exactly the string in match_exp, when match_op is set to 1 this will contain a regular expression to match.
match_len
match_len = Match Length. Allows you to match a specific length of string.
subst_exp
subst_exp = Substitute Expression. If match_op is set to 0 this will contain be empty If match_op is 1 this will contain the same as match_exp.
repl_exp
repl_exp = replacement expression. If match_op is set to 0 this will contain the string to replace the matched string.
If match_op is set to 1 this can contain the regex group matching (\1, \2, etc) and any suffixes / prefixes (for example 61\1 will prefix 61 and add the contents of matched group 1).
attrs
Attributes. Often used as a descriptive name for the matched rule.
Getting Regex Rules Setup
The real power of the dialplan comes from Regular Expression matching. Let’s look at some use cases and how to solve them with Dialplans.
Note for MySQL users: MySQL treats \ as the escape character, but we need it for things like matching a digit in Regex (that’s \d ) – So keep in mind when inserting this into MySQL you may need to escale the escape, so to enter \d into the match_exp field in MySQL you’d enter \\d – This has caught me in the past!
The hyperlinks below take you to the examples in Regex101.com so you can preview the rules and make sure it’s matching what it should prior to putting it into the database.
Speed Dial
Let’s start with a simple example of a speed dial. When a user dials 101 we want to translate it to a PSTN number of 0212341234.
Without Regex this looks very similar to our first example, we’ve just changed the dialplan id (dpid) and the match_op and repl_exp.
Once we’ve added it to the database we’ll reload the dialplan module and dump dialplan 2 to check it all looks correct:
Now let’s test what happens if we do a dialplan translate on dialplan 2 with 101.
Tip: If you’re testing a dialplan and what you’re matching is a number, add s: before it so it matches as a number, not a string.
dialplan.translate 2 s:101
Here we can see we’ve matched 101 and the output is the PSTN number we wanted to translate too.
Interoffice Dial
Let’s take a slightly more complex example. We’ve got an office with two branches, office A’s phone numbers start with 0299991000, and they have 4 digit extensions, so extension 1002 maps to 0299991002, 0299991003 maps to extension 1003, etc.
From Office B we want to be able to just dial the 4 digit extensions of a user in Office A.
This means if we receive 1003 we need to prefix 029999 + 10003.
Then another reload and translate, and we can test again.
dialplan.reload dialplan.translate 3 s:1003 (Translates to 0299991003) dialplan.translate 3 s:1101 (no translation)
Interoffice Dial Failure Route (Priorities)
So let’s say we’ve got lots of branches configured like this, and we don’t want to just get “No Translation” if a match isn’t found, but rather send it to a specific destination, say reception on extension 9000.
So we’ll keep using dpid 3 and we’ll set all our interoffice dial rules to have priority 1, and we’ll create a new entry to match anything 4 digits long and route it to the switch.
This entry will have a higher priority value than the other so will only mach if nothing else with a lower priority number matches.
Now we’ve got Group 2 containing the data we need, we just need to prefix 613 in front of it.
Let’s go ahead an put this into the database, with dialplan ID set to 4, match_op set to 1 (for regex)
Then we’ll do a dialplan reload and a dialplan dump for dialplan ID 4 to check everything is there:
Now let’s put it to the test.
dialplan.translate 4 s:0399999999
Bingo, we’ve matched the regex, and returned 613 and the output of Regex Match group 2. (999999999)
Let’s expand upon this a bit, a valid 0NSN number could also be a mobile (0400000000) or a local number in a different area code (0299999999, 0799999999 or 0899999999).
We could create a dialplan entry for each, our we could expand upon our regex to match all these scenarios.
Now let’s update the database so that once we’re matched we’ll just prefix 61 and the output of regex group 2.
Again we’ll do a dialplan reload and a dialplan dump to check everything.
Now let’s run through our examples to check they correctly translate:
And there you go, we’re matched and the 0NSN formatted number was translated to E.164.
Adding to Kamailio Routing
So far we’ve just used kamcmd’s dialplan.translate function to test our dialplan rules, now let’s actually put them into play.
For this we’ll use the function
dp_translate(id, [src[/dest]])
dp_translate is dialplan translate. We’ll feed it the dialplan id (id) and a source variable and destination variable. The source variable is the equivalent of what we put into our kamcmd dialplan.translate, and the destination is the output.
In this example we’ll rewrite the Request URI which is in variable $rU, we’ll take the output of $rU, feed it through dialplan translate and save the output as $rU (overwrite it).
Let’s start with the Speed Dial example we setup earlier, and put that into play.
if(method=="INVITE"){
xlog("rU before dialplan translation is $rU");
dp_translate("2", "$rU/$rU");
xlog("rU after dialplan translation is $rU");
}
The above example will output our $rU variable before and after the translation, and we’re using Dialplan ID 2, which we used for our speed dial example.
So let’s send an INVITE from our Softphone to our Kamailio instance with to 101, which will be translated to 0212341234.
Before we do we can check it with Kamcmd to see what output we expect:
dialplan.translate 2 s:101
Let’s take a look at the out put of Syslog when we call 101.
But our INVITE doesn’t actually go anywhere, so we’ll add it to our dispatcher example from the other day so you can see it in action, we’ll relay the INVITE to an active Media Gateway, but the $rU will change.
if(method=="INVITE"){
xlog("rU before dialplan translation is $rU");
dp_translate("2", "$rU/$rU");
xlog("rU after dialplan translation is $rU");
ds_select_dst(1, 12);
t_on_failure("DISPATCH_FAILURE");
route(RELAY);
}
Let’s take a look at how the packet captures now look:
UA > Kamailio: INVITE sip:101@kamailio SIP/2.0 Kamailio > UA: SIP/2.0 100 trying -- your call is important to us Kamailio > MG1: INVITE sip:0212341234@MG1 SIP/2.0
So as you can see we translated 101 to 0212341234 based on the info in dialplan id 2 in the database.
That’s all well and good if we dial 101, but what if we dial 102, there’s no entry in the database for 102, as we see if we try it in Kamcmd:
dialplan.translate 2 s102
And if we make a call to 102 and check syslog:
rU before dialplan translation is 102 rU after dialplan translation is 102
Let’s setup some logic so we’ll respond with a 404 “Not found in Dialplan” response if the dialplan lookup doesn’t return a result:
if(dp_translate("2", "$rU/$rU")){
xlog("Successfully translated rU to $rU using dialplan ID 2");
}else{
xlog("Failed to translate rU using dialplan ID 2");
sl_reply("404", "Not found in dialplan");
exit;
}
By putting dp_translate inside an if we’re saying “if dp_translate is successful then do {} and the else will be called if dp_translate wasn’t successful.
Let’s take a look at a call to 101 again.
UA > Kamailio: INVITE sip:101@kamailio SIP/2.0 Kamailio > UA: SIP/2.0 100 trying -- your call is important to us Kamailio > MG1: INVITE sip:0212341234@MG1 SIP/2.0
Still works, and a call to 102 (which we don’t have an entry for in the dialplan).
UA > Kamailio: INVITE sip:102@kamailio SIP/2.0 Kamailio > UA: SIP/2.0 404 Not found in dialplan
Hopefully by now you’ve got a feel for the dialplan module, how to set it up, debug it, and use it.
The Dispatcher module is used to offer load balancing functionality and intelligent dispatching of SIP messages.
Let’s say you’ve added a second Media Gateway to your network, and you want to send 75% of traffic to the new gateway and 25% to the old gateway, you’d use the load balancing functionality of the Dispatcher module.
Let’s say if the new Media Gateway goes down you want to send 100% of traffic to the original Media Gateway, you’d use the intelligent dispatching to detect status of the Media Gateway and manage failures.
These are all problems the Dispatcher Module is here to help with.
Before we get started….
Your Kamailio instance will need:
Installed and running Kamailio instance
Database configured and tables created (We’ll be using MySQL but any backed is fine)
kamcmd & kamctl working (kamctlrc configured)
Basic Kamailio understanding
The Story
So we’ve got 4 players in this story:
Our User Agent (UA) (Softphone on my PC)
Our Kamailio instance
Media Gateway 1 (mg1)
Media Gateway 2 (mg2)
Our UA will make a call to Kamailio. (Send an INVITE)
Kamailio will keep track of the up/down status of each of the media gateways, and based on rules we define pick one of the Media Gateways to forward the INVITE too.
The Media Gateways will playback “Media Gateway 1” or “Media Gateway 2” depending on which one we end up talking too.
Configuration
Parameters
You’ll need to load the dispatcher module, by adding the below line with the rest of your loadmodules:
loadmodule "dispatcher.so"
Next we’ll need to set the module specific config using modparam for dispatcher:
modparam("dispatcher", "db_url", DBURL) #Use DBURL variable for database parameters
modparam("dispatcher", "ds_ping_interval", 10) #How often to ping destinations to check status
modparam("dispatcher", "ds_ping_method", "OPTIONS") #Send SIP Options ping
modparam("dispatcher", "ds_probing_threshold", 10) #How many failed pings in a row do we need before we consider it down
modparam("dispatcher", "ds_inactive_threshold", 10) #How many sucessful pings in a row do we need before considering it up
modparam("dispatcher", "ds_ping_latency_stats", 1) #Enables stats on latency
modparam("dispatcher", "ds_probing_mode", 1) #Keeps pinging gateways when state is known (to detect change in state)
Most of these are pretty self explanatory but you’ll probably need to tweak these to match your environment.
Destination Setup
Like the permissions module, dispatcher module has groups of destinations.
For this example we’ll be using dispatch group 1, which will be a group containing our Media Gateways, and the SIP URIs are sip:mg1:5060 and sip:mg2:5060
From the shell we’ll use kamctl to add a new dispatcher entry.
You can use kamctl to show you the database entries:
kamctl dispatcher show
A restart to Kamailio will make our changes live.
Destination Status / Control
Checking Status
Next up we’ll check if our gateways are online, we’ll use kamcmd to show the current status of the destinations:
kamcmd dispatcher.list
Here we can see our two media gateways, quick response times to each, and everything looks good.
Take a note of the FLAGS field, it’s currently set to AP which is good, but there’s a few states:
AP – Active Probing – Destination is responding to pings & is up
IP – Inactive Probing – Destination is not responding to pings and is probably unreachable
DX – Destination is disabled (administratively down)
AX – Looks like is up or is coming up, but has yet to satisfy minimum thresholds to be considered up (ds_inactive_threshold)
TX – Looks like or is, down. Has stopped responding to pings but has not yet satisfied down state failed ping count (ds_probing_threshold)
Adding Additional Destinations without Restarting
If we add an extra destination now, we can add it without having to restart Kamailio, by using kamcmd:
kamcmd dispatcher.reload
There’s some sanity checks built into this, if the OS can’t resolve a domain name in dispatcher you’ll get back an error:
Administratively Disable Destinations
You may want to do some work on one of the Media Gateways and want to nicely take it offline, for this we use kamcmd again:
kamcmd dispatcher.set_state dx 1 sip:mg1:5060
Now if we check status we see MG1’s status is DX:
Once we’re done with the maintenance we could force it into the up state by replacing dx with ap.
It’s worth noting that if you restart Kamailio, or reload dispatcher, the state of each destination is reset, and starts again from AX and progresses to AP (Up) or IP (Down) based on if the destination is responding.
Routing using Dispatcher
The magic really comes down to single simple line, ds_select_dst();
The command sets the destination address to an address from the pool of up addresses in dispatcher.
You’d generally give ds_select_dst(); two parameters, the first is the destination set, in our case this is 1, because all our Media Gateway destinations are in set ID 1. The next parameter is is the algorithm used to work out which destination from the pool to use for this request.
Some common entries would be random, round robin, weight based or priority value.
In our example we’ll use a random selection between up destinations in group 1:
if(method=="INVITE"){
ds_select_dst(1, 4); #Get a random up destination from dispatcher
route(RELAY); #Route it
}
Now let’s try and make a call:
UA > Kamailio: SIP: INVITE sip:1111111@Kamailio SIP/2.0
Kamailio > UA: SIP: SIP/2.0 100 trying -- your call is important to us
And bingo, we’re connected to a Media Gateway 1. If I try it again I’ll get MG2, then MG1, then MG2, as we’re using round robin selection.
Destination Selection Algorithm
We talked a little about the different destination select algorithm, let’s dig a little deeper into the common ones, this is taken from the Dispatcher documentation:
“0” – hash over callid
“4” – round-robin (next destination).
“6” – random destination (using rand()).
“8” – select destination sorted by priority attribute value (serial forking ordered by priority).
“9” – use weight based load distribution.
“10” – use call load distribution.
“12” – dispatch to all destination in setid at once
For select destination sorted by priority (8) to work you need to include a priority, you can do this when adding the dispatcher entry or after the fact by editing the data. In the below example if MG1 is up, calls will always go to MG1, if MG1 is down it’ll go to the next highest priority (MG2).
For use weight based load distribution (9) to work, you’ll need to set a weight as well, this is similar to priority but allows you to split load, for example you could put weight=25 on a less powerful or slower destination, and weight=75 for a faster or more powerful destination, so the better destination gets 75% of traffic and the other gets 25%. (You don’t have to do these to add to 100%, I just find it easier to think of them as percentages).
use call load distribution (10) allows you to evenly split the number of calls to each destination. This could be useful if you’ve got say 2 SIP trunks with x channels on each trunk, but only x concurrent calls allowed on each. Like adding a weight you need to set a duid= value with the total number of calls each destination can handle.
dispatch to all destination in setid at once (12) allows you to perform parallel branching of your call to all the destinations in the address group and whichever one answers first will handle the call. This adds a lot of overhead, as for each destination you have in that set will need a new dialog to be managed, but it sure is quick for the user. The other major issue is let’s say I have three carriers configured in dispatcher, and I call a landline.
That landline will receive three calls, which will ring at the same time until the called party answers one of the calls. When they do the other two calls will stop ringing. This can get really messy.
Managing Failure
Let’s say we try and send a call to one of our Media Gateways and it fails, we could forward that failure response to the UA, or, better yet, we could try on another Media Gateway.
Let’s set a priority of 10 to MG1 and a priority of 5 to MG2, and then set MG1 to reject the call.
We’ll also need to add a failure route, so let’s tweak our code:
So now we’ve made a functional bare-bones PBX using Kamailio, and we’ve touched upon a few of the key functions Kamailio can do, but let’s go over them again to recap.
Future Kamailio posts I’ll be talking about using specific modules and using Kamailio for specific use cases, such as load balancing traffic between carriers and monitoring their up/down status, scaling Asterisk by front ending it with Kamailio, adding/rewriting/removing headers with Kamailio and stateful vs stateless operation, so stick around, but here’s an overview of what we’ve learned.
Routing blocks make code cleaner and allow reuse of the same blocks, you can call a route to do a function without having to write out what to do every time.
In the below example we go from the default “request_route” block, where all new messages start, and jump to the block named “RESPOND_501”.
request_route {
route(RESPOND_501); #Jump to the RESPOND_501 block
}
route[RESPOND_501]{
sl_reply("501", "Not Implemented"); #Send 501 reply
}
From RESPOND_501 we send a stateless reply to whoever sent us the message.
We introduced xlog to write data to the log, and then viewed that data in Syslog.
Building upon having saved the location of registered endpoints we looked up the locations we had Address on Record entries for and forwarded the traffic to them, to allow to make calls between registered UAs.
Quickly we saw issues with this though, if the UA we wanted to reach wasn’t registered, if we hung up before the called party answered, and a host of other scenarios,
We addressed the user not registered scenario and then talked briefly about the routing blocks that come with Kamailio and how they’ll be our savoir.
We put back the default routing blocks that come with Kamailio after highlighting the difficulties with trying to do everything yourself.
We talked about how route(RELAY); will better handle message relaying than tm_relay(); alone, and using route(WITHINDLG); to manage within dialog requests & branching.
Using a database back-end (MySQL) we setup our UAs to be authenticated when registering and before making calls, and listed our Carrier’s IPs so we’d only accept inbound calls from our carrier, not random folks online.
We used kamctl to add users and manage address groups.
route(REQINIT); was used to manage traffic validation.
auth_challenge(“$fd”, “0”); was used to authenticate UAs
and (allow_source_address(“200”)) was used to authenticate carriers.
So by now we’ve secured our box and we’re able to route calls between registered endpoints.
Next up we’ll need to add some external connectivity, meaning we can reach destinations that aren’t directly registered on our Kamailio instance.
We’ve signed up with imaginary carrier at “imaginarycarrier.com” so we can make / receive calls from the PSTN using them as a trunk. They’ll be authenticating us based on our Source IP which we’ve let them know.
At the moment, when we receive an INVITE where the destination isn’t registered, we respond with a 404:
sl_reply("404", "User not Registered"); #If looking up location fails reply with 404
But now we’ve got a carrier we can send calls to if the destination isn’t on our PBX, so we won’t need to reply 404 anymore for calls from our users.
So let’s only give the 404 reply to calls in from our carrier (inbound calls), and instead of giving a 404 response to callers from within our organisation, let’s send the calls to the carrier to make an outbound call.
This means calls to any destination that isn’t registered on Kamailio will go to the Carrier.
But we’ll need to still respond with the 404 response if a carrier sends us a call to a destination that isn’t registered, like an inbound call to a user who isn’t registered.
route[ONNETINVITE]{
if(!lookup("location")){ #Try looking up location
#If looking up location fails then:
if(allow_source_address("200")){ #Check if the request has come from a carrier
sl_reply("404", "User not Registered"); #If if it is from a carrier to a destination we don't have registered reply with 404
exit; #And exit
}else{ #If it's not from a carrier
route(TOCARRIER); #Route the call out to the carrier (to make an external call)
}
}
route(RELAY); #Relay traffic to endpoint
exit(); #Exit
}
Next we’ll need to create our TOCARRIER route,
route[TOCARRIER]{ #Route to send calls to a carrier at 192.168.200.130
rewritehost("imaginarycarrier.com"); #Rewrite host to be the carrier's IP
route(RELAY); #Route relay
}
So let’s put this together and try and make an outbound call.
Call Flow
Outbound call to Carrier
First we see our UA make the call leg to Kamailio
UA to Kamailio: SIP: INVITE sip:61299999999@kamailio SIP/2.0
Kamailio asks the UA to authenticate itself and send that again, the UA does:
It’s not in address group 200, as the from address isn’t one of our Carrier’s IPs, so it ends up at
route(TOCARRIER);
The route block itself rewrites the host part of the request to be the Carrier’s IP, and then forwards it on to the carrier.
route[TOCARRIER]{ #Route to send calls to a carrier at 192.168.200.130
rewritehost("imaginarycarrier.com"); #Rewrite host to be the carrier's IP
route(RELAY); #Route relay
}
So now the INVITE has been forwarded to imaginarycarrier.com, and because we called route(RELAY); it’ll handle all the in dialog requests.
Inbound call from Carrier
So now we know how an outbound call flows, let’s look at inbound calls from the carrier.
Carrier to Kamailio: INVITE sip:61312341234@kamailio SIP/2.0
Kamailio responds with a provisional response of 100 Trying
Kamailio to Carrier: SIP: SIP/2.0 100 trying -- your call is important to us
Now Kamailio checks to see the method type – It’s INVITE, and if the source address is in Address Group 200 (As we defined in Part 8), it is, so it calls the route(ONNETINVITE) block.
if(method=="INVITE"){
if(allow_source_address("200")){ #If from a Carrier IP
route(ONNETINVITE); #Call INVITE handling bloc
Once we get to the ONNETINVITE block Kamailio tries to lookup the location, to see if there’s a device registered with a username that matches 61312341234, using:
lookup("location")
There is, so the route(relay) is called, which forwards the INVITE to the IP it has an Address on Record for for 61312341234.
The carrier sends the INVITE to Kamailio, Kamailio calls lookup location, which fails to return a location as 61299999999 isn’t registered.
Next we check to see if the call is from a carrier by checking if the source address of the INVITE is equal to address group 200.
if(allow_source_address("200")){ #Check if the request has come from a carrier
sl_reply("404", "User not Registered"); #If if it is from a carrier to a destination we don't have registered reply with 404
exit;
As the source address is in address group 200, the carrier gets the 404 “User not Registered” reply, as we see in the packet capture:
Kamailio > Carrier: SIP: SIP/2.0 404 User not Registered
SIP was written to be fast and resonably lightweight.
At the time SIP was created in 1996, Motorola just had launched it’s first flip phone, the web was only 100,000 websites online and I was playing Pokémon.
Security wasn’t so much an afterthought, but rather not something everyone was as conscious of as they are today.
UDP is the protocol of choice for most SIP deployments, which opens it up for Message Amplification attacks.
As the world saw a few years back with DNS Amplification attacks (Good explanation of how Message Amplification works courtesy of Cloudflare), amplification attacks are enabled by DNS requests being smaller than DNS responses, and carrier networks that don’t verify the source of their traffic allowing someone to request a DNS lookup saying they’re from an IP that isn’t theirs, and that IP getting flooded.
SIP is vulnerable to this too, not exactly zero-day exploits, but something that hasn’t been looked at outside of the theoretical sphere, so I thought I’d roll up my sleeves and see how bad it can look.
For starters it’s worth remembering for a Message Amplification attack to work, it’s got to amplify. RF Engineering will teach you that amplification is the ratio of power in to power out, and it’s the same for Message Amplification, the size of the packet we send has to be smaller than the packet received, else we’re just using someone else to do our dirty work, but not amplifying.
Typical Response Sizes
These are small SIP messages I created in Python using sockets, they’re not the absolute smallest you could go, but they were as small as I could go and still get through the basic packet validation / sanity checks.
Some SIP Proxies drop traffic missing required headers while some don’t, I’ve included the required headers.
I’ve pointed the traffic at a Kamailio instance and measured the bytes sent vs bytes returned.
Method
Request Size (bytes)
Response Size (bytes)
Gain
OPTIONS
168
209
1.2x
REGISTER
380
411
1x
INVITE
197
377
1.9x
Content Length Mismatch
339
400 Missing Required Header in Request
300
Max Forwards Exceeded
213
So the best we can get is a packet 1.9 times the size of the packet we put in, which means SIP isn’t the best for Message Amplification attacks, but passable, so long as it keeps responding.
INVITE gets our best amplification and we can tune this to get the request smaller.
The Workhorse
Carrier grade SIP servers are pretty powerful machines, able to handle huge amounts of traffic, quite literally hundreds of millions a day, generally split across geographic areas and clustered, all on high quality low loss, low latency IP links.
If you have 20,000 subscribers sending a keep alive every 60 seconds, you’re at 72 million dialogs consisting of two packets each (144,000,000 SIP messages).
So after some stripping down I managed to get a valid INVITE that would be responded too with an auth challenge (407 Proxy Authentication Required) which was 125 bytes on the wire, while the response was 330, giving me a gain of 2.64 times what I put in. (I send 125 bytes, I get back 330)
The Setup
We’ve got 3 IPs we’re dealing with here,
Our victim is on 10.0.1.15. UDP port 5060 won’t even be open for this poor fellow, but he’ll get flooded.
Next is our attacker who’s machine is on 10.0.1.12, but claiming their source IP is 10.0.1.15 (the Victim’s IP)
From here our attacker will be sending SIP traffic to 10.0.1.110 (our “carrier” / SIP server), which will send it’s responses to the victim. I’ve spun up an Asterisk instance because it’s the voice eng version of sticky tape, I’d love to test this against something a Broadsoft platform, but licences are hard to come by.
I setup the Asterisk instance to be single threaded, on a box with just enough resources to run, to try this small scale.
I wrote a threaded Python script that will ramp up the number of messages exponentially, we’ll start by sending one message per second, then two messages per second, and so on.
And we’ll do this until something breaks.
The Results
In short – inconclusive at first, but kinda scary after that.
Asterisk died really quickly. “Exceptionally long queue length” popped up after the first second. Interestingly, the box eventually came good and actually replied to every one of our requests, and even sent a BYE. Cute.
So I modified the script to be a bit less aggressive, a random wait time between 0 and 1 seconds between loops for each thread.
I got about 60 seconds in before Asterisk really stopped responding to traffic.
So I tweaked my script again, enabled multi threading on the “carrier” and tried again.
So here’s the best rate (packets per second) I could get after a lot of tweaking:
Peak Receive: 14,000 packets per second Peak Send: 20,000 packets per second
In terms of packet size – what we really care about, the results were actually pretty promising:
Peak receive rate of 67Mbps, for which we were putting in ~25Mbps.
So can SIP be used for message amplification attacks? Sure.
Is it particulary practical? Not really. There’s easier targets out there for the time being, so VoIP will be spared the worst of it.
But for a carrier weaponisation of carrier SIP server should be a real fear.
Protection for Carriers
Don’t use UDP for your SIP traffic.
It’s easier said than done, I know… But the reasoning for putting SIP on UDP was primarily speed and limited bandwidth, but with more and more fibre in the ground it’s no longer the case.
SIP over TCP (better yet use it as an excuse to move to TLS), will protect you from some of these attacks.
Flood protection is built into most SBCs these days, if your box is being used to hit a specific target, the source IP will be masquerading as the target. So blocking that and not responding is your best bet. Lots of SBCs still respond with a 4xx “Rate Limiting” response instead of just dropping the traffic, ideally you’d disable the nice “Rate liming” response and just drop the traffic.
Traffic modeling, GeoIP blocking and rate limiting per IP & destination port will also help, as well as monitoring.
Ultimately you can’t stop spoofed UDP traffic coming into your network, but you can stop UDP traffic leaving your network, and if everyone did that we wouldn’t be in this mess.
UDP spoofing is made possible by networks that don’t verify that the traffic that’s leaving their network is traffic that is sourced from your network.
Your core routers know what IPs are assigned to your network, and should be configured to drop traffic that’s leaving the network but not coming from those IPs.
IETF came up with this solution, and it’s built into all major router OSes:
In our last post we went over all the theory, now let’s get started implementing these security features.
Kamailio’s core is a basis to start from, but many common needs are covered by special modules that we need to load to handle certain scenarios.
In order to authenticate traffic, we’ll need to have a source of authentication info (auth_db module) and authorization (permissions module). For this we’ll be using MySQL (although you could use text files, PostGres, etc) to store both sets of data, and using phpMyAdmin to make everything a bit more accessible.
We’ll build upon our last tutorial but we’ll install MySQL and phpMyAdmin:
apt-get install mysql-server phpmyadmin
After following along the install prompts we’ll setup our database connection.
Each of the two modules we’ll be using (auth_db and permissions) require a database source. In each we could specify our database info but instead we’ll create a new variable and fill it with our database info so we only need to update it in one place.
Let’s setup a new MySQL user for our Kamailio instance (in production you’d only grant privileges on the DB we’re going to use):
mysql> CREATE USER 'kamailio'@'localhost' IDENTIFIED BY 'my5yhtY7zPJzV8vu';
mysql> GRANT ALL PRIVILEGES ON * . * TO 'kamailio'@'localhost';
mysql> FLUSH PRIVILEGES;
We’ll now use the kamdbctl tool, bundled with Kamailio to create the database tables for us:
kamdbctl create
You’ll be asked for the root password for MySQL and if you want some optional tables (we don’t just yet) and presto, all the tables are now created!
We can now login with phpMyAdmin and see the tables we just added:
Adding Database Connectivity to Kamailio
The example config is designed to be nice and modular, so by simply un-commenting the WITH_MYSQL variable and setting the DBURL variable we’ll have set our MySQL database up for the modules we need.
We’ll change:
# *** To enable mysql:
# - define WITH_MYSQL
#
To:
# *** To enable mysql:
#!define WITH_MYSQL
#
So now we’ve defined an variabled named WITH_MYSQL
You’ll see later in the config there’s conditional (if statement) that looks at if WITH_MYSQL has been defined:
# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
# as: auth_db, acc, usrloc, a.s.o.
#!ifndef DBURL
#!define DBURL "mysql://kamailio:my5yhtY7zPJzV8vu@localhost/kamailio"
#!endif
#!endif
We’ll change the !define DBURL to include the password in the Database connection string,
It breaks up like this:
mysql:// is the database type (you could use text:// for text based DB or pgsql:// for Postgres)
First part is the username:password@host/table
In this case, our username is kamailio, our password is the one we created (my5yhtY7zPJzV8vu), our host is localhost and our table is kamailio
Adding IP Authentication & Challenge / Response Auth
Like we defined the #!define WITH_MYSQL we’ll define two other blocks to add Authentication:
# *** To enable authentication execute:
# - enable mysql
# - define WITH_AUTH
# - add users using 'kamctl'
#
# *** To enable IP authentication execute:
# - enable mysql
# - enable authentication
# - define WITH_IPAUTH
We’ll change to:
# *** To enable authentication execute:
# - enable mysql
#!define WITH_AUTH
# - add users using 'kamctl'
#
# *** To enable IP authentication execute:
# - enable mysql
# - enable authentication
#!define WITH_IPAUTH
Adding Users & IP Addresses
Now we’ve gone and added these blocks to the code we’ll go about adding some users and IP addresses, to do this we’ll use the kamctl tool.
kamctl is another tool (like kamcmd) used to modify / change the Kamailio config, it’s a shell wrapper for managing Kamailio database among other things.
We do need to setup a few things to get the kamctl working, and to do that we’ve got to edit the kamctlrtc file in the /etc/kamailio directory, to include the details of the database we just setup.
Adding Users
Now we can get to work adding some users from the command line:
kamctl add 61312341234 supersecretpassword
Here we can see adding a user with the username 61312341234 and the password supersecretpassword. These will makeup the username and password we’ll have on our SIP endpoints. (We’ll make them match the phone numbers of our trunks to make the routing easier down the track)
We’ll add another user so we can make calls between users when we’re testing later too, we’ll add them using kamctl add USERNAME PASSWORD again.
Now if we have a look in the subscriber table in phpMyAdmin we can see the users we created:
Adding Carrier IPs
Next we’ll add the IP Address our carrier is going to send us calls from, so we can allow them call our users.
Kamailio’s permissions module relies on address groups – This means we could have address group (we’ll just use the number 200), which we decide is for our carriers, and add all our carrier’s IP addresses in here, without needing to put each of them into the config.
We could create another address group 300 and put the subnets of our offices in there and only allow REGISTER messages from IPs in those groups.
We’ll go through how to use the groups later on, but for now we’ll just add a new IP address to group 200, with a /32 subnet mask, on port 5060 with the name “Carrier IP address” so we know what it is.
kamctl address add 200 10.0.1.102 32 5060 "Carrier IP address"
Now if we have a look in the permissions table in Kamailio you’ll see the info we just added.
By now you’ve probably caught on to the fact kamctl is just a command line tool to add data to the database, but it’s useful none the less.
The final step is to reload the permissions address table – This is done when we restart but it’s good to know you can update it without a restart.
kamctl address reload
Adding authentication / authorization to REGISTER messages
Now let’s actually put this all into practice, the first thing we’ll do is call the REQINIT route to make sure our traffic if (reasonably) clean and take care of the basics.
request_route {
route(REQINIT); #Call REQINIT (Request Initial) route to filter out the worst of the bad traffic and take care of the basics.
Next we’ll setup how we handle REGISTER traffic, adding an auth challenge and only saving the location if the UA successfully responds to the challenge.
if(method=="REGISTER"){ # authenticate requests
if (!auth_check("$fd", "subscriber", "1")) { #If credentials don't match what we have in Subscriber table
auth_challenge("$fd", "0"); #Send an Auth Challenge
exit; #Stop processing
}
save("location"); #Save the location as an AoR
exit; #Stop processing
}
This may all seem a bit backward, but this is an example from the Kamailio devs we’re using, so it shows “the right way” of doing this, let’s break it down.
We know our if(method==”REGISTER”) means we’ll only do this check for REGISTER messages.
The auth_check checks to see if the presented credentials in our auth header are correct. For the first INVITE we don’t have an auth header so it’s not correct, and if we have an invalid password it’s also not correct. You’ll notice it’s prefixed with a if(!auth_check) meaning this if conditional block is only called if it fails the authentication check, and if we do fail the authentication check we issue an auth_challenge to generate a 401 response with an authentication header and send it back to the UA. Then we exit (stop processing).
As the above had an exit we know we’ll only hit blocks below if our credentials are correct, otherwise we’d just get the auth_challenge and exit from the auth_check block.
So as we know these credentials are correct we’ll save the location as an address on record using the save(“location”) function and exit.
So that’s our REGISTER block now handling & requiring authentication, now after restarting Kamailio we can register SIP devices with the username and password we setup, but if we get the username or password wrong, we’ll get rejected.
We can add extra users using the kamctl add command we touched on earlier.
Authorising / Authenticating INVITE messages
INVITE messages are used to setup sessions (calls), so it’s important we secure this too. At this point we’re authenticating to REGISTER, but not create a call (INVITE).
First let’s add a simple check to see if the INVITE has come from the IP of one of the carriers we defined earlier.
For this we’ll use the allow_source_address() command to see if the source address matches what we defined earlier using kamctl address add to address group 200 in the MySQL database.
if(method=="INVITE"){
if(allow_source_address("200")){ #If from a Carrier IP
if(!lookup("location")){ #Try looking up location
sl_reply("404", "User not Registered"); #If looking up location fails reply with 404
exit; #And exit
}
t_relay(); #Relay traffic to endpoint
exit(); #Exit
}else{
sl_reply("403", "Nope. Don't know who you are");
}
}
So we’ve got a simple if for if the source address is in group 200.
Presto, this works for calls from carriers coming in to registered endpoints! We can get inbound calls.
Small catch is our users can’t dial each other any more, as their IP isn’t in address group 200, they just get the 403 “Don’t know who you are” response.
Now we could go and add the subnet where our users are located, but then there’d be no point in using passwords at all. But before we do this let’s create a new routing module, called INVITE to keep everything pretty.
At the very bottom of our config we’ll add
route[ONNETINVITE]{
if(!lookup("location")){ #Try looking up location
sl_reply("404", "User not Registered"); #If looking up location fails reply with 404
exit; #And exit
}
t_relay(); #Relay traffic to endpoint
exit(); #Exit
}
And then we’ll remove most of the code in our if(method==”INVITE”){ block and replace it with this:
if(method=="INVITE"){
if(allow_source_address("200")){ #If from a Carrier IP
route(ONNETINVITE); #Call INVITE handling bloc
}else{
sl_reply("403", "Nope. Don't know who you are");
}
}
Now we’ve made it so we just call ROUTE(INVITE);when we have an INVITE we’ve authenticated. This will save us a lot of extra code when we add our checks to see if the call is from a user we recognize, instead of running through the lookup(“location”) code and relying, we’ll just call route(ONNETINVITE); when we’re happy we know who they are and off we go.
if(method=="INVITE"){
if(allow_source_address("200")){ #If from a Carrier IP
route(ONNETINVITE); #Call INVITE handling bloc
}else{
if (!auth_check("$fd", "subscriber", "1")) { #If credentials don't match what we have in Subscriber table
auth_challenge("$fd", "0"); #Send an Auth Challenge
exit; #Stop processing
}
route(ONNETINVITE); #Call invite handling block
}
}
You may recognize the !auth_check blocks as the same code we used for authenticating REGISTER messages, we’re using it again as it’s the same auth mechanism.
If we pass it we call the route(ONNETINVITE);
If we look at the packet captures we can see our INVITE gets a 407 “Proxy Authentication required” response back from Kamailio.
And the UA then resends the INVITE with an authentication header with correct username and password and we’re on our way!
And that’s it! Phew.
In production we’d want to handle other types of messages that we’d also want to authenticate, we’ll talk about that further down the line, but keep in mind every feature you add what the security ramifications might be.
Next up we’ll use our new found sense of security to add the ability to call numbers off-net (on the PSTN) via a SIP provider!
Before we get too much further, we’ll take a breather and talk about security. We saw in the last tutorial how we’ve got to have a way to handle every scenario, like how to handle a CANCEL to an INVITE, and security is the same.
You are responsible for the code you write being secure.
In this series we’ll ultimately end up with a basic PBX with the ability to make and receive calls between registered users and via trunks from a carrier.
Soon we’ll add the ability to call numbers not registered on our system (trunks) to our system, but now is a good time to talk about securing your system, as the threat of toll fraud is very real, as well as calls from random devices waking you up in the middle of the night.
VoIP security is a big topic, there’s a whole lot of wrong ways to do things, so think before you do.
Authentication, Authorisation and Accounting
Most telephony platforms employ some form of AAA, it’s a good minimum starting point for our system to ensure we Authenticate traffic, to make sure it is who it claims to be, we Authorise traffic to make sure it’s allowed to traverse this part of the network and use theses resources, and finally we’ll Account for resources used, for example recording CDRs and cost for the services used.
Authentication
Authentication in this example we’ll using the Challenge / Response based authentication for traffic from users (Such as REGISTER and INVITE messages), as shown in this example.
When we get an INVITE or a REGISTER from a user, we’ll check their username / password matches what we’ve got on record.
We’ll use IP Address authorisation for inbound traffic from carriers (so we can make sure that we’ll only allow calls from carriers, not just any IP on the internet sending INVITES).
This means when we get an INVITE from a Carrier to send us an inbound call, we’ll make sure it’s from the Carrier’s IP address before we start our user’s phones ringing.
So we’ve got that and we lookup 1234567 in our database and get where we need to route it to:
SELECT 'forward_destination' FROM 'imaginary_route_database' WHERE 'dialed_number' = '1234567';
Seems harmless enough right?
Now imagine someone sends an INVITE that looks like this:
INVITE sip:'; DROP TABLE *;@example.com
Now let’s look at how our SQL query would look:
SELECT 'forward_destination' FROM 'imaginary_route_database' WHERE 'dialed_number' = ''; DROP TABLE *; '';
And suddenly we find all our tables have been deleted.
So it’s important we filter everything, and make sure the traffic is valid.
SQL injection is a fun example, but more common examples could include not checking the Max Forwards header and looping packets out and back in infinitely (imagine no spanning tree and plugging a switch into itself) and not responding to known bad user agents like sipcli and sipvicious.
Luckily again the Kamailio team have covered this before, Kamailio comes with a module for handling a lot of the common threats.
With the boilerplate routes we talked about in the last tutorial all you have to do is add route(REQINIT); at the start of your routing block and you’ll get:
Dodgy UA filtering (You may want to add your own)
Basics of Flood Prevention (if you want it)
max_forward checking (for handling routing loops)
OPTIONS response handling
Sanity checking
In the next post we’ll go about putting these things into practice.
In the last tutorial we saw some issues, calls hung up before they were answered (CANCEL), we also would have run into issues with timeouts, issues if the remote end was registered but no longer responding, behind a NAT etc.
We saw a real example of what I touched on on the opener to this series, that you have to deal with everything, and it’s a daunting task.
Kamailio allows you to deal with all these problems yourself, writing your own routing blocks, but it also comes with a bunch of useful routing blocks in the example config, that we can re-use so we don’t need to specify how to manage every little thing ourselves – unless we want to.
So lets add some of these useful routing blocks,
We’ll add this at the start of our request_route{ block
request_route {
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}
# handle requests within SIP dialogs
route(WITHINDLG);
So now our config looks like this:
request_route {
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}
# handle requests within SIP dialogs
route(WITHINDLG);
if(method=="INVITE"){
if(!lookup("location")){
sl_reply("404", "User not Registered");
exit;
}
lookup("location");
t_relay();
exit();
}
if(method=="REGISTER"){
save("location");
exit;
}
xlog("No idea how to respond to method $rm");
sl_reply("501", "Not Implemented");
}
As you can see we’ve added an if statement to match if the method is CANCEL or ACK, and referenced some routing blocks:
route(RELAY)
route(WITHINDLG);
We’ve also added some code to manage ACKs, but we’ll go through that on our lesson on Statefulness, for now just roll with it.
You’ll probably notice if you try and use this config that it won’t work, that’s because we’re referencing these two routing blocks without actually having defined them.
To keep us moving I’ve added all the routing blocks that come with the default Kamailio config, and added in our code to the link below;
To recap, we added the boilerplate routes that come with Kamailio and referenced them in our code to better handle in dialog responses.
This is because handling all these possible scenarios, like NAT, cancel, no response, REINVITE, UPDATE, etc, etc, would take us ages to cover, and require a pretty good understanding of Kamailio and of SIP in practice.
So use the example I’ve linked above and tune in next time, where we’ll talk about adding security and authentication to our system before we connect it to the outside world.
So now we’ve got Kamailio handling REGISTER traffic, and we know what IPs endpoints are on, so let’s join this together and let’s route a call between two endpoints via our Proxy!
We’ll work on the config we were working on in the previous tutorial:
/* 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(method=="INVITE"){
sl_reply("480", "Temporarily Unavailable");
exit;
}
if(method=="REGISTER"){
save("location");
exit;
}
sl_reply("501", "Not Implemented");
}
Let’s change how we handle the INVITE messages in our if(method==”INVITE”) block so instead of responding with a 480 Unavailable response, let’s lookup the location we saved if it was a REGISTER and forward the INVITE to the IP we’ve got.
Let’s break down each of the new functions we’ve introduced:
lookup(“location”);
In the last post we introduced save(“location”); which saves an Address on Record for the URI. lookup(“location”); looks up the IP address of the URI we’re after in the Address on Record table we wrote to with our save(“location”); call and automatically sets it as the destination IP of where we’re going to send the message.
When a user registered via a SIP REGISTER request, we saved their details, now we’re looking them up.
You could even replace the lookup(“location”) call with say a SQL lookup on an address book, and save the output to the IP the INVITE will be forwarded to, but lookup(“location”); does this it all in one function.
t_relay();
t_relay is transactional relay function. By transactional it means Kamailio remembers this session next time it’s referenced, we’ll touch upon transaction aware / stateful SIP proxies later, but for now what you need to know is it forwards the INVITE we just received to the address we got from our lookup(“location”).
exit();
Exit bails out so we won’t keep processing after that, it doesn’t just bail out of our current conditional but stops processing this request further. Without it we’d still go on and send the 501 Not Implemented reply.
The Setup
I’ve setup two phones – One using the username 106 and the other using 108, I’ve pointed both to the IP of our Kamailio instance and they’ve registered. Keep in mind if you’ve restarted Kamailio it’ll lose it’s location table, as it’s stored in memory, so you’ll have to force the phones to register again.
You can check you’ve got devices registered by using kamcmd ul.dump at the command line to confirm you’ve got entries for both. I have so I’ll try dialling 108 from 106:
Here we can see the INVITE coming in, the lookup(“location”) doing it’s magic and forwarding the messages to the IP of 108.
The 100 Trying, 180 Ringing and 200 OK messages all go through on their own as they’re part of an existing dialog which we’ll touch on why later in the series,
So we’re done here right? Well, not quite…
What if a call is made to an endpoint that isn’t registered? Or if they send a CANCEL before the call is answered?
Let’s take a look at what happens if we try this:
The INVITE is received by Kamailio, a 100 TRYING response is sent back, but Kamailio isn’t really doing anything, after about 10 seconds I try and end the call from my softphone (CANCEL) and I get a 501 response back as we haven’t defined how to handle CANCEL messages, so there’s no way to end the call, and we’re stuck…
When things don’t go to plan…
As I said in the Introduction to Kamailio post, the power in Kamailio is the ability to define how every part functions, but to quote Spiderman – With great power comes great responsibility, and Peter Parker didn’t have the 269 pages of RFC3261 to grapple with…
We know the above example works fine when there’s a device registered on the other end and if they answer, but if they don’t? As we’ve seen, we’ve got to cater for every scenario.
Handling no registered user
Instead of just lookup(“location”) to lookup location we can put it in an if statement with some code to be run if it fails.
We’ll use the the if(!lookup(“location”)){ } call to create a new conditional for if lookup location fails, and inside it we can reply with a 404 and exit.
The if(!){} block can be used to try and run a command, and if it fails, it’ll run what’s between the {}.
In our scenario we try and lookup the location, if that fails, for example because we don’t have an Address on Record for that destination, it executes what’s between the {} in this case we use sl_reply to send a 404 error back and then exit:
So now if an INVITE is received to a destination we can’t find via a lookup(“location”); we respond with a 404 and exit.
Handling CANCEL (Calls terminated before answered)
So now we’ve specified how we’ll handle calling a device that isn’t registered, next we have to specify how we deal with CANCEL messages, so if the calling party hangs up (cancels) the call before it’s answered by the called party, it stops ringing the called party.
This means our SIP server has to get the CANCEL message from the caller who’s given up, respond with it with a 200 OK to the caller, before forwarding that CANCEL onto the next hop (the user we were trying to call).
By this point you’re probably dreading your work ahead of you as you try and understand how to handle every scenario, every eventuality and all the weird gotchas like CANCEL.
If only there was someone else that had done this before…
Luckily the Kamailio team have got pretty great examples in thekamailio.cfg file that ships by default (that one we ruthlessly gutted to keep it simple in the second tutorial) and now is a good time to re-introduce some of these routing blocks, that are reused in most Kamailio instances.
The boilerplate that gets you up and running faster and means you don’t have to worry about quite as many possible scenarios or read about how to handle CANCELS ad nauseam.
In the next tutorial we’ll talk about these routing blocks, and add them to our code to manage scenarios like CANCEL, timeouts to the remote destination, and how you can use these blocks to speed things up and keep everything RFC compliant.
As we talked about in the post onSIP Registrars, SIP Registrars take the REGISTER requests from SIP endpoints and store their contact details in the form of an Address on Record (AoR). This AoR contains the URI and the endpoints’ current IP it just sent the REGISTER message from.
The primary use of this is it allow us to know how to reach people. It’s kind of like an address book for mapping current IP against a SIP URI, as IP Addresses of SIP endpoints may change often.
So our Kamailio instance is going to receive an REGISTER message, store the Contact as an address on record, and respond 200 OK. Let’s build upon the config we started with in the last tutorial:
/* 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(method=="INVITE"){
sl_reply("480", "Temporarily Unavailable");
exit;
}
sl_reply("501", "Not Implemented");
}
So let’s add an if statement to manage REGISTER messages, and save their location:
/* 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(method=="INVITE"){
sl_reply("480", "Temporarily Unavailable");
exit;
}
if(method=="REGISTER"){
save("location");
exit;
}
sl_reply("501", "Not Implemented");
}
So we’ve added an IF statement to find if the SIP method is a REGISTER message, and if it is, we’ll call the save(“location”); function.
The save() function saves the Contact address we just received to a database (in this case one in memory) in the form of a SIP URI and current Contact location, these two bits of info combined are known as an address on record (AoR) and the save function, if successful, responds with a 200 OK automatically.
I’ve pointed a SIP endpoint at it again, we revive the REGISTER, respond with 200 OK,
As you can see REGISTER sent to us, 200 OK responded.
So that’s it – Our SIP endpoint is happy and by calling the save(“location”) we’ve called the built in function to store the Contact as an Address on Record and respond with 200 OK.
So how do we access this information and what can we do with it?
Kamailio comes with two tools for accessing Kamailio while it’s running. In this example we’ll use kamcmd to check what’s registered on our system. After we’ve seen a device register, from command line we’ll run:
kamcmd ul.dump
This calls kamcmd the Kamailio command line tool, and calls the ul.dump function. ul is short for userloc – The module for user location management, and dump outputs all the contents of the userloc table.
Here we can see the info we’ve stored from the URI of the user (sip:[email protected]:60153) who sent a REGISTER at the Unix timestamp in “Last-Modified”.
So now we’ve created a SIP registrar, in the next tutorial we’ll use this information to route SIP INVITE messages to a registered endpoint, looking up it’s IP and get a call happening!
In production you’d generally not have a SIP registrar like this open to accept any registrations, you’d authenticate SIP endpoints using the REGISTER -> 401 Unauthorised -> REGISTER -> 200 OK process outlined here. To do this using the usrloc module we’d need to introduce the concept of a stateful proxy, as we need to remember what the challenge we sent them was to compare it with what we’re expecting, but we’ll touch upon this later in the series.