Building Android APN / Carrier Config

As anyone who’s setup a private LTE network can generally attest, APNs can be a real headache.

SIM/USIM cards, don’t store any APN details. In this past you may remember having to plug all these settings into your new phone when you upgraded so you could get online again.

Today when you insert a USIM belonging to a commercial operator, you generally don’t need to put APN settings in, this is because Android OS has its own index of APNs. When the USIM is inserted into the baseband module, the handset’s OS looks at the MCC & MNC in the IMSI and gets the APN settings automatically from Android’s database of APN details.

There is an option for the network to send the connectivity details to the UE in a special type of SMS, but we won’t go into that.

All this info is stored on the Android OS in apns-full-conf.xml which for non-rooted (stock) devices is not editable.

Instead the devices get updates through the OS updates which pull the latest copy of this file from Google’s Android Open Source Git repo, you can view the current master file here.

This file can override the user’s APN configuration, which can lead to some really confusing times as your EPC rejects the connection due to an unrecognized APN which is not what you have configured on the UE’s operating system, but it instead uses APN details from it’s database.

The only way around this is to change the apns-full-conf.xml file, either by modifying it per handset or submitting a push request to Android Open Source with your updated settings.

(I’ve only tried the former with rooted devices)

The XML file itself is fairly self explanatory, taking the MCC and MNC and the APN details for your network:

<apn carrier="CarrierXYZ"
      mcc="123"
      mnc="123"
      apn="carrierxyz"
      type="default,supl,mms,ims,cbs"
      mmsc="http://mms.carrierxyz.com"
      mmsproxy="0.0.0.0"
      mmsport="80"
      bearer_bitmask="4|5|6|7|8|12"
/>

Once you’ve added yours to the file, inserting the USIM, rebooting the handset or restarting the carrier app is all that’s required for it to be re-read and auto provision APN settings from the XML file.

Further reading

APN and CarrierConfig | Android Open Source Project

Carrier Configuration | Android Open Source Project

UICC Carrier Privileges | Android Open Source Project

/etc/apns-full-conf.xml – Master Branch

Kamailio Bytes – Databases

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.

As always I’ve posted my running source code for you to play with on GitHub, and I’ll do a few follow up posts on other database backends you may want to use other than MySQL.

Kamailio Bytes – xHTTP

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:

route[SEND_MESSAGE]{
        $uac_req(method)="MESSAGE";
        $uac_req(ruri)="sip:10.0.1.5:5060";
        $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)="Camera detected movement!";
        uac_req_send();

}

(I talk a bit more about the UAC module and this process in the post on UAC basics and SIP instant messaging on my post on SIMPLE)

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:

loadmodule "xhttp.so"
modparam("xhttp", "url_match", "^/sip/")

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:

And if we check syslog we should also see:

Jul 27 00:32:31 nick-voice-dev7 /usr/sbin/kamailio[111271]: ERROR: : Recieved HTTP request with request /sip/Hello%20world!</p>

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:

listen=tcp:0.0.0.0:80
listen=udp:0.0.0.0:5060

Now when we point our browser to http://your-kamailio-ip/sip/Mr.%20Watson,%20come%20here,%20I%20want%20to%20see%20you.

We’ll get this on our SIP endpoint:

Beautiful!

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!

Here’s a copy of my running code for your reference.

Kamailio Bytes – Geoip2

GeoIP2 allows simple Geo IP location parsing using mmdb files, to allow us to map IP addresses to geographic locations in standardized format.

Getting the GeoIP Data

MaxMind provide GeoIP2 formatted data ready for use, albeit with limited accuracy.

We need to download them from MaxMind and extract them for use, so let’s download the file:

#> wget https://geolite.maxmind.com/download/geoip/database/GeoLite2-City.tar.gz

Next we’ll extract the data:

#> tar -xzvf GeoLite2-City.tar.gz

Next we’ll add the below to our Kamailio config (replace the path to the GeoLite2-City.mmdb to your directory).

loadmodule "geoip2.so"
modparam("geoip2", "path", "/home/ubuntu/GeoLite2-City_20190709/GeoLite2-City.mmdb");

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.

Full example config on GitHub here.

Diameter Routing Agents (DRA)

Diameter is used extensively in 3GPP networks (Especially LTE) to provide the AAA services.

The Diameter protocol is great, and I’ve sung it’s praises before, but one issue operators start to face is that there are a lot of diameter peers, each of which needs a connection to other diameter peers.

https://en.wikipedia.org/wiki/IP_Multimedia_Subsystem#/media/File:Ims_overview.png

This diagram is an “Overview” showing one of each network element – In reality almost all network elements will exist more than once for redundancy and scalability.

What you start to end up with is a rats nest of connections, lines drawn everywhere and lots of manual work and room for human error when it comes to setting up the Diameter Peer relationships.

Let’s say you’ve got 5x MME, 5x PCRF, 2x HSS, 5x S-SCSF and 5x Packet Gateways, each needing Diameter peer relationships setup, it starts to get really messy really quickly.

Enter the Diameter Routing Agent – DRA.

Now each device only needs a connection to the DRA, which in turn has a connection to each Diameter peer. Adding a new MME doesn’t mean you need to reconfigure your HSS, just connect the MME to the DRA and away you go.

I’ll cover using Kamailio to act as a Diameter routing agent in a future post.

The Sad story of ENUM in Australia

ENUM was going to change telephone routing. No longer would you need to pay a carrier to take your calls across the PSTN, but rather through the use of DNS your handset would look up a destination and route in a peer to peer fashion.

Number porting would just be a matter of updating NAPTR records, almost all calls would be free as there’s no way/need to charge and media would flow directly from the calling party to the called party.

In 2005 ACMA became the Tier 1 provider from RIPE for the ENUM zone 4.6.e164.arpa

A trial was run and Tier 2 providers were sought to administer the system and to verify ownership of services before adding NAPTR records for individual services and referral records for ranges / delegation.

In 2007 the trial ended with only two CSPs having signed up and a half a dozen test calls made between them.

Now, over a decade later as we prepare for the ISDN switch off, NBN is almost finished rolling out, the Comms Alliance porting specs remain as rigid as ever, it might be time to look again at ENUM in Australia…

PyHSS – Python 3GPP LTE Home Subscriber Server

I recently started working on an issue that I’d seen was to do with the HSS response to the MME on an Update Location Answer.

I took some Wireshark traces of a connection from the MME to the HSS, and compared that to a trace from a different HSS. (Amarisoft EPC/HSS)

The Update Location Answer sent by the Amarisoft HSS to the MME over the S6a (Diameter) interface includes an AVP for “Multiple APN Configuration” which has the the dedicated bearer for IMS, while the HSS in the software I was working on didn’t.

After a bit of bashing trying to modify the S6a responses, I decided I’d just implement my own Home Subscriber Server.

The Diameter interface is pretty straight forward to understand, using a similar structure to RADIUS, and with the exception of the Crypto for the EUTRAN Authentication Vectors, it was all pretty straight forward.

If you’d like to know more you can download PyHSS from my GitLab page, and view my Diameter Primer post and my post on Diameter packet structure.

NBNco Australia network map

Kamailio Bytes – Routing to geo local RTPengine Instances with Kamailio

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,

We’d set it up in the way outlined in this post,

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.

Adding Multiple RTP Engines to Kamailio Database

After adding database functionality to our Kamailio instance as we covered in this post, we’ll just need to add the follow lines to our config:

loadmodule "rtpengine.so"
modparam("rtpengine", "db_url", DBURL)
modparam("rtpengine", "table_name", "rtpengine")
modparam("rtpengine", "setid_avp", "$avp(setid)")

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.

INSERT INTO `rtpengine` (`id`, `setid`, `url`, `weight`, `disabled`, `stamp`) VALUES (NULL, '6', 'udp:WA-POP.rtpengine.nickvsnetworking.com:2223', '1', '0', NOW());
INSERT INTO `rtpengine` (`id`, `setid`, `url`, `weight`, `disabled`, `stamp`) VALUES (NULL, '2', 'udp:NSW-POP.rtpengine.nickvsnetworking.com:2223', '1', '0', NOW());
INSERT INTO `rtpengine` (`id`, `setid`, `url`, `weight`, `disabled`, `stamp`) VALUES (NULL, '0', 'udp:localhost:2223', '1', '0', NOW());

We’ll restart Kamailio, and check the status of the RTPengines we added:

#> kamcmd rtpengine.show all
{
        url: udp:NSW-POP.rtpengine.nickvsnetworking.com:2223
        set: 2
        index: 1
        weight: 1
        disabled: 0
        recheck_ticks: 0
}
{
        url: udp:WA-POP.rtpengine.nickvsnetworking.com:2223
        set: 6
        index: 3
        weight: 1
        disabled: 0
        recheck_ticks: 0
}
{
        url: udp:localhost:2223
        set: 6
        index: 3
        weight: 1
        disabled: 0
        recheck_ticks: 0
}

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!

SIP Extensions – History Info

The History Info extension defined in RFC7044 sets a way for an INVITE to include where the session (call) has been before that.

For example a call may be made to a desk phone, which is forwarded (302) to a home phone. The History Info extension would add a History Info header to the INVITE to the home phone, denoting the call had come to it via the desk phone.

Here the home phone can see the call first tried [email protected], at the same time tried [email protected] and [email protected] and [email protected], base don the index values.

More Info:

https://tools.ietf.org/html/rfc7044

https://tools.ietf.org/html/rfc7131

PyRTP – Simple RTP Library for Python

I recently had a scenario where I had to encode and decode RTP packets off the wire.

I wrote a Python Library to handle it which I’ve published for anyone to use.

Encoding data is quite simple, it takes a dictionary of values to fill the headers and payload and returns hex data to be sent down the wire:

payload = 'd5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5' 

packet_vars = {'version' : 2, 'padding' : 0, 'extension' : 0, 'csi_count' : 0, 'marker' : 0, 'payload_type' : 8, 'sequence_number' : 306, 'timestamp' : 306, 'ssrc' : 185755418, payload' : payload} 

PyRTP.GenerateRTPpacket(packet_vars)             #Generates hex to send down the wire 

And decoding is the same but reverse, feed it hex data and it returns a dict of values:

packet_bytes = '8008d4340000303c0b12671ad5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5'

rtp_params = PyRTP.DecodeRTPpacket(packet_bytes) #Returns dict of values from packet

Hopefully it’ll save someone else some time in the future.

For more info on RTP see:

RTP – More than you Wanted to Know for a deep dive into the packet structure

Diameter Packet Structure

We talked a little about what the Diameter protocol is, and how it’s used, now let’s look at the packets themselves.

Each Diameter packet has at a the following headers:

Version

This 1 byte field is always (as of 2019) 0x01 (1)

Length

3 bytes containing the total length of the Diameter packet and all it’s contained AVPs.

This allows the receiver to know when the packet has ended, by reading the length and it’s received bytes so far it can know when that packet ends.

Flags

Flags allow particular parameters to be set, defining some possible options for how the packet is to be handled by setting one of the 8 bits in the flags byte, for example Request Set, Proxyable, Error, Potentially Re-transmitted Message,

Command Code

Each Diameter packet has a 3 byte command code, that defines the method of the request,

The IETF have defined the basic command codes in the Diameter Base Protocol RFC, but many vendors have defined their own command codes, and users are free to create and define their own, and even register them for public use.

3GPP have defined a series of their own command codes.

Application ID

To allow vendors to define their own command codes, each command code is also accompanied by the Application ID, for example the command code 257 in the base Diameter protocol translates to Capabilities Exchange Request, used to specify the capabilities of each Diameter peer, but 257 is only a Capabilities Exchange Request if the Application ID is set to 0 (Diameter Base Protocol).

If we start developing our own applications, we would start with getting an Application ID, and then could define our own command codes. So 257 with Application ID 0 is Capabilities Exchange Request, but command code 257 with Application ID 1234 could be a totally different request.

Hop-By-Hop Identifier

The Hop By Hop identifier is a unique identifier that helps stateful Diameter proxies route messages to and fro. A Diameter proxy would record the source address and Hop-by-Hop Identifier of a received packet, replace the Hop by Hop Identifier with a new one it assigns and record that with the original Hop by Hop Identifier, original source and new Hop by Hop Identifier.

End-to-End Identifier

Unlike the Hop-by-Hop identifier the End to End Identifier does not change, and must not be modified, it’s used to detect duplicates of messages along with the Origin-Host AVP.

AVPs

The real power of Diameter comes from AVPs, the base protocol defines how to structure a Diameter packet, but can’t convey any specific data or requests, we put these inside our Attribute Value Pairs.

Let’s take a look at a simple Diameter request, it’s got all the boilerplate headers we talked about, and contains an AVP with the username.

Here we can see we’ve got an AVP with AVP Code 1, containing a username

Let’s break this down a bit more.

AVP Codes are very similar to the Diameter Command Codes/ApplicationIDs we just talked about.

Combined with an AVP Vendor ID they define the information type of the AVP, some examples would be Username, Session-ID, Destination Realm, Authentication-Info, Result Code, etc.

AVP Flags are again like the Diameter Flags, and are made up a series of bits, denoting if a parameter is set or not, at this stage only the first two bits are used, the first is Vendor Specific which defines if the AVP Code is specific to an AVP Vendor ID, and the second is Mandatory which specifies the receiver must be able to interpret this AVP or reject the entire Diameter request.

AVP Length defines the length of the AVP, like the Diameter length field this is used to delineate the end of one AVP.

AVP Vendor ID

If the AVP Vendor Specific flag is set this optional field specifies the vendor ID of the AVP Code used.

AVP Data

The payload containing the actual AVP data, this could be a username, in this example, a session ID, a domain, or any other value the vendor defines.

AVP Padding

AVPs have to fit on a multiple of a 32 bit boundary, so padding bits are added to the end of a packet if required to total the next 32 bit boundary.

Diameter Basics

3GPP selected Diameter protocol to take care of Authentication, Authorization, and Accounting (AAA).

It’s typically used to authenticate users on a network, authorize them to use services they’re allowed to use and account for how much of the services they used.

In a EPC scenario the Authentication function takes the form verifying the subscriber is valid and knows the K & OP/OPc keys for their specific IMSI.

The Authorization function checks to find out which features, APNs, QCI values and services the subscriber is allowed to use.

The Accounting function records session usage of a subscriber, for example how many sessional units of talk time, Mb of data transferred, etc.

Diameter Packets are pretty simple in structure, there’s the packet itself, containing the basic information in the headers you’d expect, and then a series of one or more Attribute Value Pairs or “AVPs”.

These AVPs are exactly as they sound, there’s an attribute name, for example username, and a value, for example, “Nick”.

This could just as easily be for ordering food; we could send a Diameter packet with an imaginary command code for Food Order Request, containing a series of AVPs containing what we want. The AVPs could belike Food: Hawian Pizza, Food: Garlic Bread, Drink: Milkshake, Address: MyHouse.
The Diameter server could then verify we’re allowed to order this food (Authorization) and charge us for the food (Accounting), and send back a Food Order Response containing a series of AVPs such as Delivery Time: 30 minutes, Price: $30.00, etc.

Diameter packets generally take the form of a request – response, for example a Capabilities Exchange Request contains a series of AVPs denoting the features supported by the requester, which is sent to a Diameter peer. The Diameter peer then sends back a Capabilities Exchange Response, containing a series of AVPs denoting the features that it supports.

Diameter is designed to be extensible, allowing vendors to define their own type of AVP and Diameter requests/responses and 3GPP have defined their own types of messages (Diameter Command Codes) and types of data to be transferred (AVP Codes).

LTE/EPC relies on Diameter and the 3GPP/ETSI defined AVP / Diameter Packet requests/responses to form the S6a Interface between an MME and a HSS, the Gx Interface between the PCEF and the PCRF, Cx Interface between the HSS and the CSCF, and many more interfaces used for Authentication in 3GPP networks.

Qos in LTE (4G) – ARP

ARP in LTE is not the Ethernet standard for address resolution, but rather the Allocation and Retention Policy.

A scenario may arise where on a congested cell another bearer is requested to be setup.

The P-GW, S-GW or eNB have to make a decision to either drop an existing bearer, or to refuse the request to setup a new bearer.

The ARP value is used to determine the priority of the bearer to be established compared to others,

For example a call to an emergency services number on a congested cell should drop any other bearers so the call can be made, thus the request for bearer for the VoLTE call would have a higher ARP value than the other bearers and the P-GW, S-GW or eNB would drop an existing bearer with a lower ARP value to accommodate the new bearer with a higher ARP value.

ARP is only used when setting up a new bearer, not to determine how much priority is given to that bearer once it’s established (that’s defined by the QCI).

QoS in LTE (4G) – MBR/AMBR/APN-MBR

MBR stands for Maximum Bit Rate, and it defines the maximum rate traffic can flow between a UE and the network.

It can be defined on several levels:

MBR per Bearer

This is the maximum bit rate per bearer, this rate can be exceeded but if it is exceeded it’s QoS (QCI) values for the traffic peaking higher than the MBR is back to best-effort.

AMBR

Aggregate Maximum Bit Rate – Maximum bit rate of all Service Data Flows / Bearers to and from the network from a single UE.

APN-MBR

The APN-MBR allows the operator to set a maximum bit rate per APN, for example an operator may choose to limit the MBR for subscriber on an APN for a MVNO to give it’s direct customers a higher speed.

(This is only applied to Non-GBR bearers)

QoS in LTE (4G) – QCI

The QCI (Quality Class Indicator) is a value of 0-9 to denote the service type and the maximum delays, packet loss and throughput the service requires.

Different data flows have different service requirements, let’s look at some examples:

A VoLTE call requires low latency and low packet loss, without low latency it’ll be impossible to hold a conversation with long delays, and with high packet loss you won’t be able to hear each other.

On the other hand a HTTP (Web) browsing session will be impervious to high latency or packet loss – the only perceived change would be slightly longer page load times as lost packets are resent and added delay on load of a few hundred ms.

So now we understand the different requirements of data flows, let’s look at the columns in the table above so we can understand what they actually signify:

GBR

Guaranteed Bit Rate bearers means our eNB will reserve resource blocks to carry this data no matter what, it’ll have those resource blocks ready to transport this data.

Even if the data’s not flowing a GBR means the resources are reserved even if nothing is going through them.

This means those resource blocks can’t be shared by other users on the network. The Uu interface in the E-UTRAN is shared between UEs in time and frequency, but with GBR bearers parts of this can be reserved exclusively for use by that traffic.

Non-GBR

With a Non-GBR bearer this means there is no guaranteed bit rate, and it’s just best effort.

Non-GBR traffic is scheduled onto resource blocks when they’re not in use by other non-GBR traffic or by GBR traffic.

Priority

The priory value is used for preemption by the PCRF.

The lower the value the more quickly it’ll be processed and scheduled onto the Uu interface.

Packet Delay Budget

Maximum allowable packet delay as measured from P-GW to UE.

Most of the budget relates to the over-the-air scheduling delays.

The eNB uses the QCI information to make its scheduling decisions and packet prioritisation to ensure that the QoS requirements are met on a per-EPS-bearer basis.

(20ms is typically subtracted from this value to account for the radio propagation delay on the Uu interface)

Packet Error Loss Rate (PELR)

This is packets lost on the Uu interface, that have been sent but not confirmed received.

The PELR is an upper boundary for how high this can go, based on the SDUs (IP Packets) that have been processed by the sender on RLC but not delivered up to the next layers (PDCP) by the receiver.

(Any traffic bursting above the GBR is not counted toward the PELR)

(The list is now larger than 0-9 with 3GPP adding extra QCI values for MCPTT, V2X, etc, the full list is available here in table 6.1.7A)

QoS in LTE (4G) – GBR & Non-GBR Bearers

GBR is a confusing concept at the start when looking at LTE but it’s actually kind of simple when we break it down.

GBR stands for Guaranteed Bit Rate, meaning the UE is guaranteed a set bit rate for the bearer.

The default bearer is always a non-GBR bearer, with best effort data rates.

Let’s look at non-GBR bearers to understand the need for GBR bearers:

As the Uu (Air) interface is shared between many UEs, each is able to transfer data. Let’s take an example of a cell with two UEs in it and not much bandwidth available.

If UE1 and UE2 are both sending the same amount of data it’ll be evenly split between the two.

But if UE1 starts sending a huge amount of data (high bit rate) this will impact on the other UEs in the cells ability to send data over the air as it’s a shared resource.

So if UE2 needs to send a stream of small but important data over the air interface, while UE2 is sending huge amounts of data, we’d have a problem.

To address this we introduce the concept of a Guaranteed Bit Rate. We tell the eNB that the bearer carrying UE2’s small but important data needs a Guaranteed Bit Rate and it reserves blocks on the air interface for UE2’s data.

So now we’ve seen the need for GBR there’s the counter point – the cost.

While UE1 can still continue sending but the eNB will schedule fewer resource blocks to it as it’s reserved some for UE2’s data flow.

If we introduced more and more UEs each requiring GBR bearers, eventually our non-GBR traffic would simply not get through, so GBR bearers have to be used sparingly.

Note: IP data isn’t like frame relay or circuit switched data that’s consistent, bit rate can spike and drop away all the time. GBR guarantees a minimum bit rate, which is generally tuned to the requirements of the data flow. For example a GBR for a Voice over IP call would reserve enough for the media (RTP stream) but no more, so as not to use up resources it doesn’t need.

NAT solutions used in VoIP

NAT is still common in Voice networks, and while we’re all awaiting the full scale adoption of IPv6, it’s still going to be a thing for some time.

I thought I’d dive into some of the NAT “solutions” that are currently in use.

Old RFC 3489 Definitions

These were the first NAT implementations used, and are still often used today.

Full cone NAT

A request from a private address is mapped to a public address and a publicly available port.

Traffic can be sent from any external device to this public address / port combination, and will be sent the internal device.

This is often statically setup, where you’d log into your router and put a NAT rule saying “Traffic on Port 5060 I want forwarded to my desk phone on 192.168.1.2” for example, and is sometimes just called a “Port forward”.

This can work fine if you’ve just got one unchanging internal address, but starts to have issues with multiple devices or dynamically assigned IPs.

Restricted Cone NAT

A request from a private address is mapped to a public address.

Traffic sent to this public address from an allowed IP will be routed to the internal device, regardless of port used.

Port Restricted Cone

Like restricted cone but only a single port may be used, traffic sent to any other port will not be routed to the internal device.

Symmetric NAT

Each request to an external destination gets a unique Public IP / Port combination to be used only by that destination, and each new request with a different source port on the internal side, or different destination on the external side, sets up a new NAT path.

RFC 5389 NAT Definitions

Endpoint Independent Mapping

Each request to an external destination gets the same public IP address / Port combination used for the outbound traffic.

Return traffic from the external destination is routed based on the source address, to the internal IP of the originating user.

It’s possible to have multiple internal devices communicating with multiple external destinations, using the same public IP address / port combination for each of them.

The source IP address of the traffic back from the external destination is used to map the path back to the internal IP.

This is efficient (doesn’t need to keep using outbound ports on the public IP) but means that it’ll only work to the requested external destination’s IP.

If you register to a SIP server on one IP, and media comes in on another, an Endpoint Independent Mapping NAT will see you with one-way audio.

Address Dependant Mapping

Each request to an external destination gets a unique public IP address / port combination used for outbound traffic.

It is reused for packets sent to the same destination, regardless of which destination port is used.

Address and Port Dependant Mapping

Same as Address Dependant Mapping but a new mapping is created for each destination and port.

Numbering Systems in Australia: E.164 vs 0-NDC-SN

You’ll often see numbers listed in different formats, which often leads to confusion.

Australian SIP networks may format numbers in either 0NDC-SN or E.164 format, leading some confusion. There’s no “correct” way, ACMA format in 0-NDC-SN, while most Australian tier 1 carriers store the records in E.164 format.

There’s no clear standard, so it’s always best to ask.

Let’s say my number is in Melbourne and is 9123 4567,

This could be expressed in Subscriber Number (SN) format:

9123 4567

The problem is a caller from Perth calling that number wouldn’t get through to me, there’s a good chance they’d get a totally unrelated business.

To stop this we can add the National Destination Code (NDC), for Victoria / Tasmania this is 3, however when dialling domestically a 0 is prefixed.

The leading 0 is a carry over from the days of step-by-step based switching, which had technical and physical design constraints that dictated the dialling plan we see today, which I’ll do a post about another day.

So to put it in 0-NDC-S format we’d list

03 9123 4567

But an international caller wouldn’t be able to reach this from their home country, they’d need to add the Country Code (CC) which for Australia is 61, so they’d dial the CC-NDC-SN

So they’d dial 61 3 9123 4567

This formatting is called E.164, defined by the ITU in The international public telecommunication numbering plan,

Sometimes this is listed with the plus symbol in front of it, like

+61 3 9123 4567

Each country has it’s own international dialling prefix, and the plus symbol is to be replaced by the international dialling prefix used in the calling country. In Australia, we replace the + with 0011, but it’s different from country to country.

RF Planning with Forsk Atoll - Importing environmental data

Forsk Atoll – WMS Map Tiles

A hack I found useful to add Google Maps / Google Satelite View / Bing Maps / Bing Arial / Open Street Maps in Forsk Atoll.

Close Atoll,

Go to C -> Program Files -> Atoll

Edit the file named atoll.ini

Paste the following into it:

[OnlineMaps]
Name1 = OpenStreetMap Standard Map
URL1 = http://a.tile.openstreetmap.org/%z/%x/%y.png
Name2 = MapQuest Open Aerial
URL2 = http://otile1.mqcdn.com/tiles/1.0.0/sat/%z/%x/%y.jpg
Name3 = 2Gis
URL3 = http://static.maps.api.2gis.ru/1.0?c...z&size=256,256
Name4 = 2Gis without logo
URL4 = http://tile2.maps.2gis.com/tiles?x=%x&y=%y&z=%z&v=37 
Name5 = Bing Aerial
URL5 = http://ecn.t3.tiles.virtualearth.net.../a%q.jpg?g=392
Name6 = Bing Hybrid
URL6 = http://ecn.t3.tiles.virtualearth.net.../h%q.jpg?g=392
Name7 = Bing Road
URL7 = http://ecn.t3.tiles.virtualearth.net.../r%q.jpg?g=392
Name8 = Yandex Road
URL8 = http://static-maps.yandex.ru/1.x/?ll...=%z&l=map&key=
Name9 = Yandex Aerial
URL9 = http://static-maps.yandex.ru/1.x/?ll...=%z&l=sat&key=
Name10 = Yandex Hybrid
URL10 = http://static-maps.yandex.ru/1.x/?ll...l=sat,skl&key=
Name11 = ArcGIS
URL11 = http://services.arcgisonline.com/Arc...e/%z/%y/%x.png
Name12 = opencyclemap
URL12 = http://tile.opencyclemap.org/cycle/%z/%x/%y.png
Name13 = Google Terrain
URL13 = http://mt.google.com/vt/lyrs=t&hl=en&x=%x&y=%y&z=%z
Name14 = Google Map
URL14 = http://mt.google.com/vt/lyrs=m&hl=en&x=%x&y=%y&z=%z
Name15 = Google Hybrid (Map + Terrain)
URL15 = http://mt.google.com/vt/lyrs=p&hl=en&x=%x&y=%y&z=%z
Name16 = Google Hybrid (Map + Satellite)
URL16 = http://mt.google.com/vt/lyrs=y&hl=en&x=%x&y=%y&z=%z
Name17 = Google Satellite
URL17 = http://mt.google.com/vt/lyrs=m&hl=en&x=%x&y=%y&z=%z
Name18 = Google Scheme
URL18 = http://mt.google.com/vt/lyrs=h&hl=en&x=%x&y=%y&z=%z
Name19 = Google Scheme2 
URL19 = http://mt.google.com/vt/lyrs=r&hl=en&x=%x&y=%y&z=%z

Save and open Atoll,

Open the Geo Tab,

Right click on Online Maps, click “New”

Select the map source (In this example I’m using OSM) & hit Ok.

Enable the Online Map layer by ticking the layer.

Bam, done.

RF Planning with Forsk Atoll - Laying out environmental data