Tag Archives: VoIP

HTTP PUT call recordings from FreeSWITCH

The mod_httapi in FreeSWITCH allows you to upload your call recordings to a HTTP server, in my case I’ve put together a Flask based Python server for a project I’m working on, which when the call ends, uploads to my web server. Presto!

<action application="record" data="http://(file=recording.wav,name=part1.PCMU,method=POST)your_domain.com/recording_upload"/>

Obviously you’ll need to replace the URL etc, but you can then just extract the POSTed file out and boom, you don’t need to store any recordings on each FreeSWITCH instance.

This is fantastic if you’re running multiple instances in a cluster or containerized, and want every FreeSWITCH instance to be dumb and with access to the same data as every other instance.

VoIP is an only child – ‘Gotchas’ on running VoIP applications inside Containers

It’s 2021, and everyone loves Containers; Docker & Kubernetes are changing how software is developed, deployed and scaled.

And yet so much of the Telco world still uses bare metal servers and dedicated hardware for processing.

So why not use Containers or VMs more for VoIP applications?

Disclaimer – When I’m talking VoIP about VoIP I mean the actual Voice over IP, that’s the Media Stream, RTP, the Audio, etc, not the Signaling (SIP). SIP is fine with Containers, it’s the media that has a bad time and that this post focuses on,

Virtualization Fundamentals

Once upon a time in Development land every application ran on it’s own server running in a DC / Central Office.

This was expensive to deploy (buying servers), operate (lots of power used) and maintain (lots of hardware to keep online).

Each server was actually sitting idle for a large part of the time, with the application running on it only using a some of the available resources some of the time.

One day Virtualization came and suddenly 10 physical servers could be virtualized into 10 VMs.

These VMs still need to run on servers but as each VM isn’t using 100% of it’s allocated resources all the time, instead of needing 10 servers to run it on you could run it on say 3 servers, and even do clever things like migrate VMs between servers if one were to fail.

VMs share the resources of the server it’s running on.

A server running VMs (Hypervisor) is able to run multiple VMs by splitting the resources between VMs.

If a VM A wants to run an operation at the same time a VM B & VM C, the operations can’t be run on each VM at the same time* so the hypervisor will queue up the requests and schedule them in, typically based on first-in-first out or based on a resource priority policy on the Hypervisor.

This is fine for a if VM A, B & C were all Web Servers.
A request coming into each of them at the same time would see the VM the Hypervisor schedules the resources to respond to the request slightly faster, with the other VMs responding to the request when the hypervisor has scheduled the resources to the respective VM.

VoIP is an only child

VoIP has grown up on dedicated hardware. It’s an only child that does not know how to share, because it’s never had to.

Having to wait for resources to be scheduled by the Hypervisor to to VM in order for it to execute an operation is fine and almost unnoticeable for web servers, it can have some pretty big impacts on call quality.

If we’re running RTPproxy or RTPengine in order to relay media, scheduling delays can mean that the media stream ends up “bursty”.

RTP packets needing relaying are queued in the buffer on the VM and only relayed when the hypervisor is able to schedule resources, this means there can be a lot of packet-delay-variation (PDV) and increased latency for services running on VMs.

VMs and Containers both have this same fate, DPDK and SR-IOV assist in throughput, but they don’t stop interrupt headaches.

VMs that deprive other VMs on the same host of resources are known as “Noisy neighbors”.

The simple fix for all these problems? There isn’t one.

Each of these issues can be overcome, dedicating resources, to a specific VM or container, cleverly distributing load, but it is costly in terms of resources and time to tweak and implement, and some of these options undermine the value of virtualization or containerization.

As technology marches forward we have scenarios where Kubernetes can expose FPGA resources to pass them through to Pods, but right now, if you need to transcode more than ~100 calls efficiently, you’re going to need a hardware device.

And while it can be done by throwing more x86 / ARM compute resources at the problem, hardware still wins out as cheaper in most instances.

Sorry, no easy answers here…

Kamailio Bytes – OnReply Route

So far with most of our discussions about Kamailio we’ve talked about routing the initial SIP request (INVITE, REGISTER, SUBSCRIBE, etc), but SIP is not a one-message protocol, there’s a whole series of SIP messages that go into a SIP Dialog.

Sure the call may start with an INVITE, but there’s the 180 RINGING, the 200 OK and the ACK that go into getting the call actually established, and routing these in-dialog messages is just as important as routing the first INVITE.

When we’ve talked about SIP routing it’s all happened in the request_route {} block:

request_route {
        xlog("Received $rm to $ru - Forwarding");
        append_hf("X-Proxied: You betcha\r\n");
        #Forward to new IP
        forward("192.168.1.110");
}

In the example above we statelessly forward any initial requests to the IP 192.168.1.110.

All the routing from that point on happens using the standard RFC3261 in-dialog routing using the Route headers.

We can add an onreply_route{} block to handle any replies from 192.168.1.110 back to the originator.

But why would we want to?

Some simple answers would be to do some kind of manipulation to the message – say to strip a Caller ID if CLIP is turned off, or to add a custom SIP header containing important information, etc.

onreply_route{
        xlog("Got a reply $rs");
        append_hf("X-Proxied: For the reply\r\n");
}

Let’s imagine a scenario where the destination our SIP proxy is relaying traffic to (192.168.1.110) starts responding with 404 error.

We could detect this in our onreply_route{} and do something about it.

onreply_route{
        xlog("Got a reply $rs");
        if($rs == 404) {
                #If remote destination returns 404
                xlog("Got a 404 for $rU");
                #Do something about it
        }
}

In the 404 example if we were using Dispatcher it’s got easily accessed logic to handle these scenarios a bit better than us writing it out here, but you get the idea.

There are a few other special routes like onreply_route{}, failure routes and event routes, etc.

Hopefully now you’ll have a better idea of how and when to use onreply_route{} in Kamailio.

Kamailio Bytes – Gotchas with Kamailio as an Asterisk Load Balancer

How do I make Kamailio work with Asterisk?

It’s a seemingly simple question, the answer to which is – however you want, sorry if that’s not a simple answer.

I’ve talked about the strengths and weaknesses of Kamailio and Asterisk in my post Kamailio vs Asterisk, so how about we use them to work together?

The State of Play

So before we go into the nitty gritty, let’s imagine we’ve got an Asterisk box with a call queue with Alice and Bob in it, set to ring those users if they’re not already on a call.

Each time a call comes in, Asterisk looks at who in the queue is not already on a call, and rings their phone.

Now let’s imagine we’re facing a scenario where the single Asterisk box we’ve got is struggling, and we want to add a second to share the load.

We do it, and now we’ve got two Asterisk boxes and a Kamailio load balancer to split the traffic between the two boxes.

Now each time a call comes in, Kamailio sends the SIP INVITE to one of the two Asterisk boxes, and when it does, that Asterisk box looks at who is in the queue and not already on a call, and then rings their phone.

Let’s imagine a scenario where a Alice & Bob are both on calls on Asterisk box A, and another call comes in this call is routed to Asterisk box B. Asterisk box B looks at who is in the queue and who is already on a call, the problem is Alice and Bob are on calls on Asterisk box A, so Asterisk box B doesn’t know they’re both on a call and tries to ring them.

We have a problem.

Scaling stateful apps is a real headache,

So have a good long hard think about how to handle these issues before going down this path!

FreeSWITCH + ESL = Programmable Voice

No great secret, I’m a big Python fan.

Recently I’ve been working on a few projects with FreeSWITCH, and looking at options for programmatically generating dialplans, instead of static XML files.

Why not Static XML?

So let’s say I define a static XML dialplan.

It works great, but if I want to change the way a call routes I need to do it from the dialplan,

That’s not ideal if you’re using a distributed cluster of FreeSWITCH instances, or if you want to update on the fly.

Static XML means we have to define our dialplan when setting up the server, and would have to reconfigure the server to change it.

So what about mod_xml_curl?

When I’ve done this in the past I’ve relied on the mod_xml_curl module.

mod_xml_curl gets the XML dialplan using Curl from a web server, and so you setup a web server using Flask/PHP/etc, and dynamically generate the dialplan when the call comes in.

This sounds good, except you can’t update it on the fly.

mod_xml_curl means call routing decisions are made at the start of the call, and can’t be changed midway through the call.

So what’s ESL?

ESL is the Event Socket Library, essentially a call comes in, an ESL request is made to an external server.

For each step in the dialplan, an ESL request will be sent to the external server which tells it to do,

ESL allows us to use all FreeSWITCH’s fantastic modules, without being limited as to having to perform the call routing logic in FreeSWITCH.

So how do I use ESL?

You’ll need two create an ESL server,

Luckily there’s premade examples for popular languages;

Once you’ve got a basic server defined we’ll need to put some config in our XML Dialplan to say “transfer all your thinking to ESL!”;

<!-- Send everything that's numbers to ESL for Processing --> 
    <extension name="esl_route">
      <condition field="destination_number" expression="^\d*$">
        <action application="socket" data="10.0.1.252:5000"/>
        <action application="log" data="ERR Made it past ESL - Play error."/>
        <action application="playback" data="ivr/ivr-call_cannot_be_completed_as_dialed"/>
      </condition>
    </extension>

In the above example my ESL server is on 10.0.1.252 / port 5000.

Now any calls coming in will be transfered to ESL and where it goes from there, is something you define in your prefered programming language.

In the next post I’ll cover how I’ve been addressing this using Python and Greenswitch.

SIP Register – Lesser Known Features

In the past we’ve covered what a SIP Registrar does, how to build one, and covered some misconceptions about what being Registered means, but there’s a few little-utilized features of SIP Registration that are quite useful.

A lot of people think there’s a one-to-one relationship between a registration Address on Record, and a username.

That doesn’t have to be the case, there are some platforms that only allow a single registration for a single username, but the RFC itself allows multiple registrations for a single username.

REGISTER requests add, remove, and query bindings.

A REGISTER request can add a new binding between an address-of-record and one or more contact addresses.

Registration on behalf of a particular address-of-record can be performed by a suitably authorized third party.

A client can also remove previous bindings or query to determine which bindings are currently in place for an address-of-record.

RFC 3261 – 10.2

Let’s say you’ve got a SIP phone on your desk at the office and at home.

What we could do is create a different username and password for home & work, and then setup some time based forward rules to ring the office from 9-5 and home outside of that.

You could register both with the same username and password, and then unplug the one at home before you leave to work, get to work, plug in your office phone, unplug it before you leave to go home, and when you get home plug back in your home phone, or if multi-device registration is supported, register both and have incoming calls ring on both.

Admittedly, platforms that support this are the exception, not the rule, but the RFC does allow it.

The other little known feature in SIP Registration is that you can query the SIP Registrar to get the list of Addresses on Record.

So there you go, factoids about SIP REGISTER method!

Kamailio Bytes – Multiple Kamailio Instances on a Single Box

For whatever reason you might want to run multiple Kamailio instances on the same machine.

In my case I was working on an all-in-one IMS box, which needed the P-CSCF, I-CSCF and S-CSCF all in the one box.

init.d File

As you probably already know, all the startup scripts for each service/daemon live in the /etc/init.d directory.

We’ll start by copying the existing init.d file for kamailio:

cp /etc/init.d/kamailio /etc/init.d/kamailio1

Next up we’ll edit it to reflect the changes we want made.

You only really need to change the DEFAULTS= parameter, but you may also want to update the description, etc.

DEFAULTS=/etc/default/kamailio1

The CFGFILE parameter we can update later in the defaults file or specify here.

Defaults File

Next up we’ll need to create a defaults file where we specify how our instance will be loaded,

Again, we’ll copy an existing defaults file and then just edit it.

cp /etc/default/kamailio /etc/default/kamailio1

The file we just created from the copy will need to match the filename we specified in the init.d file for DEFAULTS=

In my case the filename is kamailio1

In here I’ll need to at minimum change the CFGFILE= parameter to point to the config file for the Kamailio instance we’re adding.

In this case the file is called kamailio1.cfg in /etc/kamailio/

For some Ubuntu systems you’re expected to reload the daemons:

systemctl daemon-reload

Starting / Running

Once you’ve done all this you can now try and start your instance using /etc/init.d/kamailio1 start

For my example startup failed as I haven’t created the config file for kamailio1.cfg

So I quickly created a config file and tried to start my service:

/etc/init.d/kamailio1 restart

And presto, my service is running,

I can verify all is running through ps aux:

ps aux | grep kamailio1

Just keep in mind if you want to run multiple instances of Kamailio, you can’t have them all bound to the same address / port.

This also extends to tools like kamcmd which communicate with Kamailio over a socket, again you’d need to specify unique ports for each instance.

Kamailio Bytes – SIP over TLS (SIPS)

It’s probably pretty evident to most why you’d want to use TLS these days,

SIP Secure – aka sips has been around for a long time and is supported by most SIP endpoints now.

Kamailio supports TLS, and it’s setup is very straightforward.

I’ve got a private key and certificate file for the domain nickvsnetworking.com so I’ll use those to secure my Kamailio instance by using TLS.

I’ll start by copying both the certificate (in my case it’s cert.pem) and the private key (privkey.pem) into the Kamailio directory. (If you’ve already got the private key and certificate on your server for another application – say a web server, you can just reference that location so long as the permissions are in place for Kamailio to access)

Next up I’ll open my Kamailio config (kamailio.cfg), I’ll be working with an existing config and just adding the TLS capabilities, so I’ll add this like to the config:

!define WITH_TLS

That’s pretty much the end of the configuration in kamailio.cfg, if we have a look at what’s in place we can see that the TLS module loads it’s parameters from a separate file;

#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/etc/kamailio/tls.cfg")
#!endif

So let’s next jump over to the tls.cfg file and add our certificate and private key;

[server:default]
method = TLSv1
verify_certificate = yes
require_certificate = yes
certificate = fullchain.pem
private_key = privkey.pem

Boom, as simple as that,

After restarting Kamailio subscribers can now contact us via TLS using sips.

You may wish to disable TCP & UDP transport in favor of only TLS.

A note about CAs…

If you’re planning on rolling out SIP over TLS (sips) to existing IP phones it’s worth looking at what Certificate Authorities (CAs) are recognised by the IP phones.

As TLS relies on a trust model where a CA acts kind of like a guarantor to the validity of the certificate, if the IP phone doesn’t recognise the CA, it may see the certificate as Invalid.

Some IP phones for example won’t recognize Let’s Encrypt certificates as valid, while others may not recognize any of the newer CAs.

Vendor Yealink publishes a list of CAs their IP phones recognize, which could save you a lot of headaches when setting this up and buying certificates.

VoLTE Logo on Samsung Galaxy Handset

Things I wish I knew about setting up private VoLTE Networks

I’ve been working for some time on open source mobile network cores, and one feature that has been a real struggle for a lot of people (Myself included) is getting VoLTE / IMS working.

Here’s some of the issues I’ve faced, and the lessons I learned along the way,

Sadly on most UEs / handsets, there’s no “Make VoLTE work now” switch, you’ve got a satisfy a bunch of dependencies in the OS before the baseband will start sending SIP anywhere.

Get the right Hardware

Your eNB must support additional bearers (dedicated bearers I’ve managed to get away without in my testing) so the device can setup an APN for the IMS traffic.

Sadly at the moment this rules our Software Defined eNodeBs, like srsENB.

In the end I opted for a commercial eNB which has support for dedicated bearers.

ISIM – When you thought you understood USIMs – Guess again

According to the 3GPP IMS docs, an ISIM (IMS SIM) is not a requirement for IMS to work.

However in my testing I found Android didn’t have the option to enable VoLTE unless an ISIM was present the first time.

In a weird quirk I found once I’d inserted an ISIM and connected to the VoLTE network, I could put a USIM in the UE and also connect to the VoLTE network.

Obviously the parameters you can set on the USIM, such as Domain, IMPU, IMPI & AD, are kind of “guessed” but the AKAv1-MD5 algorithm does run.

Getting the APN Config Right

There’s a lot of things you’ll need to have correct on your UE before it’ll even start to think about sending SIP messaging.

I was using commercial UE (Samsung handsets) without engineering firmware so I had very limited info on what’s going on “under the hood”. There’s no “Make VoLTE do” tickbox, there’s VoLTE enable, but that won’t do anything by default.

In the end I found adding a new APN called ims with type ims and enabling VoLTE in the settings finally saw the UE setup an IMS dedicated bearer, and request the P-CSCF address in the Protocol Configuration Options.

Also keep in mind on Android at least, what you specify as your APN might be ignored if your UE thinks it knows best – Thanks to the Android Master APN Config – which guesses the best APN for you to use, which is a useful feature to almost any Android user, except the very small number who see fit to setup their own network.

Get the P-GW your P-CSCF Address

If your P-GW doesn’t know the IP of your P-CSCF, it’s not going to be able to respond to it in the Protocol Configuration Options (PCO) request sent by the UE with that nice new bearer for IMS we just setup.

There’s no way around Mutual Authentication

Coming from a voice background, and pretty much having RFC 3261 tattooed on my brain, when I finally got the SIP REGISTER request sent to the Proxy CSCF I knocked something up in Kamailio to send back a 200 OK, thinking that’d be the end of it.

For any other SIP endpoint this would have been fine, but IMS Clients, nope.

Reading the specs drove home the same lesson anyone attempting to setup their own LTE network quickly learns – Mutual authentication means both the network and the UE need to verify each other, while I (as the network) can say the UE is OK, the UE needs to check I’m on the level.

For anyone not familiar with the intricacies of 3GPP USIM Network Authentication, I’ve written about Mutual Network Authentication in this post.

In the end I added Multimedia Authentication support to PyHSS, and responded with a Crypto challenge using the AKAv1-MD5 auth,

For anyone curious about what goes on under the hood with this, I wrote about how the AKAv1-MD5 Authentication algorithm works in this post,

I saw my 401 response go back to the UE and then no response. Nada.

This led to my next lesson…

There’s no way around IPsec

According to the 3GPP docs, support for IPsec is optional, but I found this not to be the case on the handsets I’ve tested.

After sending back my 401 response the UE looks for the IPsec info in the 401 response, then tries to setup an IPsec SA and sends ESP packets back to the P-CSCF address.

Even with my valid AKAv1-MD5 auth, I found my UE wasn’t responding until I added IPsec support on the P-CSCF, hence why I couldn’t see the second REGISTER with the Authentication Info.

After setting up IPsec support, I finally saw the UE’s REGISTER with the AKAv1-MD5 authentication, and was able to send a 200 OK.

For some more info on ESP, IPsec SAs and how it works between the UE and the P-CSCF there’s a post on that too.

Get Good at Mind Reading (Or an Engineering Firmware)

To learn all these lessons took a long time,

One thing I worked out a bit late but would have been invaluable was cracking into the Engineering Debug options on the UEs I was testing with.

Samsung UEs feature a Sysdump utility that has an IMS Debugging tool, sadly it’s only their for carriers doing IMS interop testing.

After a bit of work I detailed in this post – Reverse Engineering Samsung Sysdump Utils to Unlock IMS Debug & TCPdump on Samsung Phones – I managed to create a One-Time-Password generator for this to generate valid Samsung OTP keys to unlock the IMS Debugging feature on these handsets.

I outlined turning on these features in this post.

This means without engineering firmware you’re able to pull a bunch of debugging info off the UE.

If you’ve recently gone through this, are going through this or thinking about it, I’d love to hear your experiences.

I’ll be continuing to share my adventures here and elsewhere to help others get their own VoLTE networks happening.

If you’re leaning about VoLTE & IMS networks, or building your own, I’d suggest checking out my other posts on the topic.

VoLTE / IMS – P-CSCF Assignment

The Proxy-Call Session Control Function is the first network element a UE sends it’s SIP REGISTER message to, but how does it get there?

To begin with our UE connects as it would normally, getting a default bearer, an IP address and connectivity.

Overview

If the USIM has an ISIM application on it (or IMS is enabled on the UE using USIM for auth) and an IMS APN exists on the UE for IMS, the UE will set up another bearer in addition to the default bearer.

This bearer will carry our IMS traffic and allow QoS to be managed through the QCI values set on the bearer.

While setting up the bearer the UE requests certain parameters from the network in the Protocol Configuration Options element, including the P-CSCF address.

When setting up the bearer the network responds with this information, which if supported includes the P-CSCF IPv4 &/or IPv6 addresses.

The Message Exchange

We’ll start assuming the default bearer is in place & our UE is configured with the APN for IMS and supports IMS functionality.

The first step is to begin the establishment of an additional bearer for the IMS traffic.

This is kicked off through the Uplink NAS Transport, PDN Connectivity Request from the UE to the network. This includes the IMS APN information, and the UE’s NAS Payload includes the Protocol Configuration Options element (PCO), with a series of fields the UE requires responses from the network. including DNS Server, MTU, etc.

In the PCO the UE also includes the P-CSCF address request, so the network can tell the UE the IP of the P-CSCF to use.

If this is missing it’s because either your APN settings for IMS are not valid, or your device doesn’t have IMS support or isn’t enabling it.(that could be for a few reasons).

Protocol Configuration Options (Unpopulated) used to request information from the Network by the UE

The MME gets this information from the P-GW, and the network responds in the E-RAB Setup Request, Activate default EPS bearer Context Request and includes the Protocol Configuration Options again, this time the fields are populated with their respective values, including the P-CSCF Address;

Once the UE has this setup, the eNB confirms it’s setup the radio resources through the E-RAB Setup Response.

One the eNB has put the radio side of things in place, the UE confirms the bearer assignment has completed successfully through the Uplink NAS Transport, Activate default EPS Bearer Accept, denoting the bearer is now in place.

Now the UE has the IP address(s) of the P-CSCF and a bearer to send it over, the UE establishes a TCP socket with the address specified in the P-CSCF IPv4 or IPv6 address, to start communicating with the P-CSCF.

The SIP REGISTER request can now be sent and the REGISTRATION procedure can begin.

I’ve attached a PCAP of the full exchange here.

I’ve written a bit about the Gm REGISTER procedure and how IPsec is implemented between the UE and the P-CSCF in this post.

If you’re leaning about VoLTE & IMS networks, or building your own, I’d suggest checking out my other posts on the topic.

Automated SIP testing with sipcmd

I wrote about some tests I ran with SIPp to load test the transcoding abilities of RTPengine a while back.

While SIPp allows you to create complex & powerful scenarios, sipcmd’s simple usage makes it great for quickly testing stuff.

Installation

Install prerequisites

apt-get install libopal-dev sip-dev libpt-dev libssl1.0-dev

Next up clone the GitHub repo and compile:

git clone https://github.com/tmakkonen/sipcmd
cd sipcmd
make 

To be able to call sipcmd from anywhere, copy the binary to /usr/sbin/

cp sipcmd /usr/sbin/

Usage

Unlike SIPp, sipcmd has a much more simple syntax to allow you to follow basic call scenarios, like call a destination, wait a set time and then hangup, or answer an incoming call and send a DTMF digit and wait for the called party to hangup.

So let’s get the most basic thing we can set, SIP Registration and Authentication.

sipcmd -P sip -u "nick" -c "mypassword" -w "192.168.190.129"

Now sipcmd will register on that host (192.168.190.129) with username nick and password mypassword.

And it works!

Next we’ll add a basic call scenario, call 123 wait 2 seconds (2000 ms) and then hangup.

“c123;w2000;h”

./sipcmd -P sip -u "nick" -c "mypassword" -w "192.168.190.129" -x "c123;w2000;h"

And there you have it, simple as that, we’ve made a test call, waited a set time and then hung up.

We can even combine this with monitoring / NMS systems like Nagios to run tests against the network continually.

For more advanced scenarios I’d recommend using SIPp, but for simple testing, particularly from a command line, sipcmd is a simple easy place to start.

SIP SIMPLE – Instant Messaging with SIP

People think SIP they think VoIP & phone calls, but SIP it’s the Phone Call Initiation Protocol it’s the Session Initiation Protocol – Sure VoIP guys like me love SIP, but it’s not just about VoIP.

Have you sent an SMS on a modern mobile phone recently? Chances are you sent a SMS over SIP using SIP MESSAGE method.

So let’s look a bit at SIP SIMPLE, the catchily titled acronym translates to Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (Admittedly less catchy in it’s full form).

There’s two way SIP SIMPLE can be used to implement Instant Messaging, Paging Mode with each message sent as a single transaction, and Session Mode where a session is setup between users and IMs exchanged with the same Call ID / transaction.

I’m going to cover the Paging Mode implementation because it’s simpler easier to understand.

Before we get too far this is another example of confusing terminology, let’s just clear this up; According to the RFC any SIP request is a SIP Message, like a SIP OPTIONS message, a SIP INVITE message. But the method of a SIP INVITE message is INVITE, the method of a SIP OPTIONS message is OPTIONS. There’s a SIP MESSAGE method, meaning you can send a SIP MESSAGE message using the MESSAGE method. Clear as mud? I’ll always refer to the SIP Method in Capitals, like MESSAGE, INVITE, UPDATE, etc.

The SIP MESSAGE Method

The basis of using SIP for instant messaging relies on the MESSAGE method, laid out in RFC 3428.

The SIP MESSAGE method looks / acts very similar to a SIP INVITE, in that it’s got all the standard SIP headers, but also a Message Body, in which our message body lives (funny about that), typically we’ll send messages using the Content-Type: text/plain to denote we’re sending a plaintext message.

Example MESSAGE Message Flow

Like a SIP OPTIONS Method, the MESSAGE method is simply answered with a 200 OK (No Ack).

Let’s have a look at how the MESSAGE message looks:

MESSAGE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP user1pc.domain.com;branch=z9hG4bK776sgdkse
Max-Forwards: 70
From: sip:[email protected];tag=49583
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 MESSAGE
Content-Type: text/plain
Content-Length: 18

Hello world.

After receiving the SIP MESSAGE message, the recipient simply sends back a 200 OK with the same Call-ID.

Simple as that.

You can read more about the SIP MESSAGE method in RFC 3428.

I used the SIP MESSAGE method in a Kamailio Bytes example recently where I sent a MESSAGE to an IP phone when a HTTP GET was run against Kamailio, and again to send an alert when an emergency services destination was called.

Kamailio Bytes – UAC for Remote User Registration to external SIP Server (Originating SIP REGISTER)

I’ve talked about using the UAC module, but as promised, here’s how we can use the UAC module to send SIP REGISTER requests to another SIP server so we can register to another SIP proxy.

Let’s say we’re using Kamailio to talk to a SIP Trunk that requires us to register with them so they know where to send the calls. We’d need to use Kamailio UAC module to manage SIP Registration with our remote SIP Trunk.

But Kamailio’s a proxy, why are we sending requests from it? A proxy just handles messages, right?
Proxies don’t originate messages, it’s true, and Kamailio can be a proxy, but with the UAC module we can use Kamailio as a Client instead of a server. Keep in mind Kamailio is what we tell it to be.

Getting Started

Before we can go spewing registrations out all over the internet we need to start by getting a few things in place;

First of which is configuring UAC module, which is something I covered off in my last post,

We’ll also need to have a database connection in place, again I’ve covered off connecting to a MySQL database in Kamailio here.

Once we’ve got that done we’ll need to tell the UAC module our IP Address for the from address for our Contact field, and the database URL of what we’ve setup.

modparam("uac", "reg_contact_addr", "192.168.1.99:5060")
modparam("uac", "reg_db_url", "mysql://kamailio:kamailiorw@localhost/kamailio")

I haven’t used a variable like DBURL for the database information, but you could.

Finally a restart will see these changes pushed into Kamailio.

/etc/init.d/kamailio restart

This is the end of the Kamailio config side of things, which you can find on my GitHub here.

Defining the Registration parameters

Once we’ve got a database connection in place and UAC module loaded, then we can configure an entry in the uacreg table in the database, in my example I’m going to be registering to an Asterisk box on 192.168.1.205, so I’ll insert that into my database:

mysql> INSERT INTO `uacreg` VALUES (NULL,'myusername','myusername','192.168.1.205','myusername','192.168.1.205','asterisk','myusername','mypassword','','sip:192.168.1.205:5060',60,0,0);

Note: If you’re using a later version of Kamailio (5.4+) then the DB schema changes and you may want something like this:

insert into uacreg values ('', 'myusername', 'myusername', 'mydomain', 'myusername', 'mydomain', 'asteriskrealm', 'myusername', 'mypassword', '', 'sip:remoteproxy.com:5060', 60, 0, 0, 0)

Having a look at the fields in our table makes it a bit clearer as to what we’ve got in place, setting flags to 0 will see Kamailio attempt registration. Make sure the auth_proxy is a SIP URI (Starts with sip:) and leave the auth_ha1 password empty as we haven’t calculated it.

mysql> SELECT * FROM 'uacreg' \G
            id: 2
        l_uuid: myusername
    l_username: myusername
      l_domain: 192.168.1.205
    r_username: myusername
      r_domain: 192.168.1.205
         realm: asterisk
 auth_username: myusername
 auth_password: mypassword
      auth_ha1:
    auth_proxy: sip:192.168.1.205:5060
       expires: 60
         flags: 0
     reg_delay: 0

Putting it into Play

After we’ve got our database connection in place, UAC module configured and database entries added, it’s time to put it into play, we’ll use Kamcmd to check it’s status:

kamcmd> uac.reg_reload
kamcmd> uac.reg_dump

Unfortunately from Kamcmd we’re not able to see registration status, but Sngrep will show us what’s going on:

From Sngrep we can see the REGISTRATION going out, the authentication challenge and the 200 OK at the end.

Make sure you’ve got your Realm correct, otherwise you may see an error like this:

RROR: {2 10 REGISTER [email protected]} uac [uac_reg.c:946]: uac_reg_tm_callback(): realms do not match. requested realm: [localhost]

If you’re not familiar with the SIP Registration process now’s a good time to brush up on it by having a read of my post here. – “What is a SIP Registrar?”

Kamailio Bytes – SIP UAC Module to act as a UAC / SIP Client

Kamailio is a great SIP proxy, but sometimes you might want to see requests originate from Kamailio.

While this isn’t typical proxy behaviour, RFC definitions of a proxy and technical requirements are often two different things. The UAC module allows us to use Kamailio to act as a User Agent Client instead of just a UAS.

There’s one feature I won’t cover in this post, and that’s initiating and outbound SIP Registration using the UAC module, that will get a post of it’s own in the not to distant future.

You may already be sort of using Kamailio is a UAC, if you’re using Dispatcher and sending SIP Pings, then Kamailio is sending SIP OPTIONS messages to the dispatcher destinations. If you’re using the NAT module and sending Keepalives, then you’re also using Kamailio as a UAC. The only difference is the Dispatcher and NAT Helper modules do this for us, and we’re going to originate our own traffic.

There’s a bit of a catch here, when Kamailio receives a request it follows a set of logic and does something with that request. We’re going to remain constrained by this for our example, just to keep things simple.

So let’s work on an example, if a user on our network dials a call to an emergency services number, we’ll send a text message to my IP phone to let me know who’s dialed the emergency services number.

So to start with we’ll need to load the Kamailio UAC module, using LoadModule as we would with any other module:

loadmodule "uac.so"

If you’re working on the default config file that ships with Kamailio you’ll probably have to change how record routing is handled to support UAC,

modparam("rr", "append_fromtag", 1)

Now we should have UAC support added in Kamailio, I’m going to do a bare bones example of the routing logic below, but obviously if you wanted to put this into practice in real life you’d want to actually route the SIP INVITE to an emergency services destination.

First we’ll need to find if the request is an INVITE with the Request URI to an emergency services number, I’ve programmed this in with the Australian emergency services numbers:

if(is_method("INVITE") && ($rU == "000" or $tU == "112" or $tU == "116")){      
  #Matches any INVITEs with the Request URI to Address as 000, 112 or 116
  xlog("Emergency call from $fU to $rU (Emergency number) CSeq is $cs ");
}

Now calls to 000, 112 or 116 will see the alert apear in Xlog:

07:22:41 voice-dev3 /usr/sbin/kamailio[10765]: ERROR: : Emergency call from Test to 112 (Emergency number)

So next up we need to handle the sending a SIP MESSAGE request to my IP phone on the IP 10.0.1.5 – You’re probably thinking we could use the Registrar module to lookup my registered IP address, and you’re right, but to keep things simple I’m just hardcoding it in.

So to keep our routing neat we’ll send calls to the route route(“EmergencyNotify”); and so the demo works I’ll send back a 200 OK and exit – In real life you’d want to handle this request and forward it onto emergency services.

if(is_method("INVITE") && ($rU == "000" or $tU == "112" or $tU == "116")){      
#Matches any INVITEs with the Request URI to Address as 000, 112 or 116
  xlog("Emergency call from $fU to $rU (Emergency number) CSeq is $cs ");
  route("EmergencyNotify");
  #You obviously would want this to route to an emergency services destination...
  sl_reply("200", "ok");
  exit;
}

if(is_method("INVITE")){                                                                                
  #Matches everything else
  xlog("Just a regular call from $fU to $rU");
}

Obviously we need to now create a route called route[“EmergencyNotify”]{ } where we’ll put our UAC logic.

For the UAC module we need to craft the SIP Request we’re going to send; we’re going to be sending a SIP MESSAGE request,

route["EmergencyNotify"]{
  xlog("Emergency Notify Route");
  $uac_req(method)="MESSAGE";
  $uac_req(ruri)="sip:10.0.1.5:5060";
  $uac_req(furi)="sip:Emergency Alert";
  $uac_req(turi)="sip:thisphone";
  $uac_req(callid)=$(mb{s.md5});
  $uac_req(hdrs)="Subject: Emergency Alert\r\n";
  $uac_req(hdrs)=$uac_req(hdrs) + "Content-Type: text/plain\r\n";
  $uac_req(body)="Emergency call from " + $fU + " on IP Address " + $si + " to " + $rU + " (Emergency Number)";
  $uac_req(evroute)=1;
  uac_req_send();
}

So now we’ve sort of put it all together, when a call comes into an emergency destination, like 000, the route EmergencyNotify is called which sends a SIP MESSAGE request to my IP Phone to alert me.

When a caller dials 000 I can see Kamailio sends a SIP MESSAGE to my IP Phone:

Let’s have a look at how this looks on my IP Phone:

I’ve fleshed out the code a little more to handle SIP REGISTER requests etc, and put the full running code on GitHub which you can find here.

IMS / VoLTE IPsec on the Gm Interface

For most Voice / Telco engineers IPsec is a VPN technology, maybe something used when backhauling over an untrusted link, etc, but voice over IP traffic is typically secured with TLS and SRTP.

IMS / Voice over LTE handles things a bit differently, it encapsulates the SIP & RTP traffic between the UE and the P-CSCF in IPsec Encapsulating Security Payload (ESP) payloads.

In this post we’ll take a look at how it works and what it looks like.

It’s worth noting that Kamailio recently added support for IPsec encapsulation on a P-CSCF, in the IMS IPSec-Register module. I’ll cover usage of this at a later date.

The Message Exchange

The exchange starts off looking like any other SIP Registration session, in this case using TCP for transport. The UE sends a REGISTER to the Proxy-CSCF which eventually forwards the request through to a Serving-CSCF.

This is where we diverge from the standard SIP REGISTER message exchange. The Serving-CSCF generates a 401 Unauthorized response, containing an authentication challenge in the WWW-Authenticate header, and also a Ciphering Key & Integrity Key (ck= and ik=) also in the WWW-Authenticate header.

The Serving-CSCF sends the Proxy-CSCF the 401 response it created. The Proxy-CSCF assigns a SPI for the IPsec ESP to use, a server port and client port and indicates the used encryption algorithm (ealg) and algorithm to use (In this case HMAC-SHA-1-96.) and adds a new header to the 401 Unauthorized called SecurityServer header to share this information with the UE.

The Proxy-CSCF also strips the Ciphering Key (ck=) and Integrity Key (ik=) headers from the SIP authentication challenge (WWW-Auth) and uses them as the ciphering and integrity keys for the IPsec connection.

Finally after setting up the IPsec server side of things, it forwards the 401 Unauthorized response onto the UE.

Upon receipt of the 401 response, the UE looks at the authentication challenge.

Keep in mind that the 3GPP specs dictate that IMS / VoLTE authentication requires mutual network authentication meaning the UE authenticates the network as well as the network authenticating the UE. I’ve written a bit about mutual network authentication in this post for anyone not familiar with it.

If the network is considered authenticated by the UE it generates a response to the Authentication Challenge, but it doesn’t deliver it over TCP. Using the information generated in the authentication challenge the UE encapsulates everything from the network layer (IPv4) up and sends it to the P-CSCF in an IPsec ESP.

Communication between the UE and the P-CSCF is now encapsulated in IPsec.

Wireshark trace of IPsec IMS Traffic between UE and P-CSCF

If you’re leaning about VoLTE & IMS networks, or building your own, I’d suggest checking out my other posts on the topic.

Kamailio Bytes – KEMI Intro

When learning to use Kamailio you might find yourself thinking about if you really want to learn to write a Kamailio configuration file, which is another weird scripting language to learn to achieve a task.

Enter KEMI – Kamailio Embedded Interface. KEMI allows you to abstract the routing logic to another programing language. In layman’s terms this means you can write your routing blocks, like request_route{}, reply_route{}, etc, in languages you already know – like Lua, JavaScript, Ruby – and my favorite – Python!

Why would you use KEMI?

Write in a language you already know;

You don’t need to learn how to do write complex routing logic in Kamailio’s native scripting language, you can instead do it in a language you’re already familiar with, writing your Routing Blocks in another programming language.

Change Routing on the Fly;

By writing the routing logic in KEMI allows you to change your routing blocks without having to restart Kamailio, something you can’t do with the “native” scripting language – This means you can change your routing live.

Note: This isn’t yet in place for all languages – Some still require a restart.

Leverage your prefered language’s libraries;

While Kamailio’s got a huge list of modules to interface with a vast number of different things, the ~200 Kamailio modules don’t compare with the thousands of premade libraries that exist for languages like Python, Ruby, JavaScript, etc.

Prerequisites

We’ll obviously need Kamailio installed, but we’ll also need the programming language we want to leverage setup (fairly obvious).

Configuring Kamailio to talk to KEMI

KEMI only takes care of the routing of SIP messages inside our routing blocks – So we’ve still got the Kamailio cfg file (kamailio.cfg) that we use to bind and setup the service as required, load the modules we want and configure them.

Essentially we need to load the app for the language we use, in this example we’ll use app_python3.so and use that as our Config Engine.

loadmodule "app_python3.so"
modparam("app_python3", "load", "/etc/kamailio/kemi.py")
cfgengine "python"

After that we just need to remove all our routing blocks and create a basic Python3 script to handle it,

We’ll create a new python file called kemi.py

## Kamailio - equivalent of routing blocks in Python
import sys
import Router.Logger as Logger
import KSR as KSR

# global function to instantiate a kamailio class object
# -- executed when kamailio app_python module is initialized
def mod_init():
    KSR.info("===== from Python mod init\n");
    return kamailio();


# -- {start defining kamailio class}
class kamailio:
    def __init__(self):
        KSR.info('===== kamailio.__init__\n');


    # executed when kamailio child processes are initialized
    def child_init(self, rank):
        KSR.info('===== kamailio.child_init(%d)\n' % rank);
        return 0;


    # SIP request routing
    # -- equivalent of request_route{}
    def ksr_request_route(self, msg):
        KSR.info("===== request - from kamailio python script\n");
        KSR.info("===== method [%s] r-uri [%s]\n" % (KSR.pv.get("$rm"),KSR.pv.get("$ru")));

So that’s it! We’re running,

The next step is of course, putting some logic into our Python script, but that’s a topic for another day, which I’ve covered in this post.

Running code for kamailio.cfg (Kamailio config) and kemi.py (Python3 script).

Using Wireshark to peer inside IPsec ESP VoLTE data from the P-CSCF

IPsec ESP can be used in 3 different ways on the Gm interface between the Ue and the P-CSCF:

  • Integrity Protection – To prevent tampering
  • Ciphering – To prevent inception / eavesdropping
  • Integrity Protection & Ciphering

On Wireshark, you’ll see the ESP, but you won’t see the payload contents, just the fact it’s an Encapsulated Security Payload, it’s SPI and Sequence number.

By default, Kamailio’s P-CSCF only acts in Integrity Protection mode, meaning the ESP payloads aren’t actually encrypted, with a few clicks we can get Wireshark to decode this data;

Just open up Wireshark Preferences, expand Protocols and jump to ESP

Now we can set the decoding preferences for our ESP payloads,

In our case we’ll tick the “Attempt to detect/decode NULL encrypted ESP payloads” box and close the box by clicking OK button.

Now Wireshark will scan through all the frames again, anything that’s an ESP payload it will attempt to parse.

Now if we go back to the ESP payload with SQN 1 I showed a screenshot of earlier, we can see the contents are a TCP SYN.

Now we can see what’s going on inside this ESP data between the P-CSCF and the UE!

As a matter of interest if you can see the IK and CK values in the 401 response before they’re stripped you can decode encrypted ESP payloads from Wireshark, from the same Protocol -> ESP section you can load the Ciphering and Integrity keys used in that session to decrypt them.

If you’re leaning about VoLTE & IMS networks, or building your own, I’d suggest checking out my other posts on the topic.

SIP Supported & Require

On top of plain vanilla RFC3261, there’s a series of “Extension” methods added to SIP to expand it’s functionality, common extension methods are INFO, MESSAGE, NOTIFY, PRACK and UPDATE. Although now commonplace, of these is not defined in RFC3261 so is considered an “extension” to SIP.

It’s worth just pausing here to reiterate we’re not talking extensions like in a PBX context, like extra phones, we’re talking extensions like you’d add to a house, like extra functionality.

A SIP client can request functionality from a server (UAC to a UAS), if the server does not have support for that functionality, it can reject the session on those grounds and send back a response indicating it doesn’t know how to handle that extension, like a 420 Bad ExtensionBad SIP Protocol Extension used, not understood by the server. Response.

So clients can determine what functionality a server doesn’t support if it rejects the request, but there was no way to see what functionality the server does support, and what functionality the client requires.

Enter the Supported header, initially drafted by Rosenberg & Schulzrinne in 2000, it made it into the SIP we know today (SIP v2 / RFC3261).

If a UAC or UAS requires support for an extension – For example a Media Gateway has to understand PRACK, it can use the Require header to specify the request should be rejected if support for the listed extensions is not provided.

These headers are most commonly seen in SIP OPTIONS requests.

Kamailio Bytes – Dispatcher States

You may already be familiar with Kamailio’s Disptacher module, if you’re not, you can learn all about it in my Kamailio Bytes – Dispatcher Module post.

One question that’s not as obvious as it perhaps should be is the different states shown with kamcmd dispatcher.list command;

So what do the flags for state mean?

The first letter in the flag means is the current state, Active (A), Inactive (I) or Disabled (D).

The second letter in the flag means monitor status, Probing (P) meaning actively checked with SIP Options pings, or Not Set (X) denoting the device isn’t actively checked with SIP Options pings.

AP Actively Probing – SIP OPTIONS are getting a response, routing to this destination is possible, and it’s “Up” for all intents and purposes.

IPInactively Probing – Destination is not meeting the threshold of SIP OPTIONS request responses it needs to be considered active. The destination is either down or not responding to all SIP OPTIONS pings. Often this is due to needing X number of positive responses before considering the destination as “Up”.

DX Disabled & Not Probing – This device is disabled, no SIP OPTIONS are sent.

AX Active & Not Probing– No SIP OPTIONS are sent to check state, but is is effectively “Up” even though the remote end may not be reachable.