Recently I’ve been doing some work with FreeSWITCH as an IMS Conference Factory, I’ve written a bit about it before in this post on using FreeSWITCH with the AMR codec.
Pretty early on in my testing I faced a problem with subsequent in-dialog responses, like re-INVITEs used for holding the calls.
Every subsequent message, was getting a “420 Bad Extension” response from FreeSWITCH.
The ReINVITEThe 420 Bad Extension Response
So what didn’t it like and why was FreeSWITCH generating 420 Bad Extension Responses to these subsequent messages?
Well, the “Extensions” FreeSWITCH is referring to are not extensions in the Telephony sense – as in related to the Dialplan, like an Extension Number to identify a user, but rather the Extensions (as in expansions) to the SIP Protocol introduced for IMS.
The re-INVITE contains a Require header with sec-agree which is a SIP Extension introduced for IMS, which FreeSWITCH does not have support for, and the re-INVITE says is required to support the call (Not true in this case).
Using a Kamailio based S-CSCF means it is easy to strip these Headers before forwarding the requests onto the Application Server, which is what I’ve done, and bingo, no more errors!
In iOS 15, Apple added support for iPhones to support SMS over IMS networks – SMSoIP. Previously iPhone users have been relying on CSFB / SMSoNAS (Using the SGs interface) to send SMS on 4G networks.
Getting this working recently led me to some issues that took me longer than I’d like to admit to work out the root cause of…
I was finding that when sending a Mobile Termianted SMS to an iPhone as a SIP MESSAGE, the iPhone would send back the 200 OK to confirm delivery, but it never showed up on the screen to the user.
The GSM A-I/F headers in an SMS PDU are used primarily for indicating the sender of an SMS (Some carriers are configured to get this from the SIP From header, but the SMS PDU is most common).
The RP-Destination Address is used to indicate the destination for the SMS, and on all the models of handset I’ve been testing with, this is set to the MSISDN of the Subscriber.
But some devices are really finicky about it’s contents. Case in point, Apple iPhones.
If you send a Mobile Terminated SMS to an iPhone, like the one below, the iPhone will accept and send back a 200 OK to this request.
The problem is it will never be displayed to the user… The message is marked as delivered, the phone has accepted it it just hasn’t shown it…
SMS reports as delivered by the iPhone (200 OK back) but never gets displayed to the user of the phone as the RP-Destination Address header is populated
The fix is simple enough, if you set the RP-Destination Address header to 0, the message will be displayed to the user, but still took me a shamefully long time to work out the problem.
RP-Destination Address set to 0 sent to the iPhone, this time it’ll get displayed to the user.
After getting AMR support in FreeSWITCH I set about creating an IMS Application Server for VoLTE / IMS networks using FreeSWITCH.
So in IMS what is an Application Server? Well, the answer is almost anything that’s not a CSCF.
An Application Server could handle your Voicemail, recorded announcements, a Conference Factory, or help interconnect with other systems (without using a BGCF).
I’ll be using mine as a simple bridge between my SIP network and the IMS core I’ve got for VoLTE, with FreeSWITCH transcoding between AMR to PCMA.
Setting up FreeSWITCH
You’ll need to setup FreeSWITCH as per your needs, so that’s however you want to use it.
This post won’t cover setting up FreeSWITCH, there’s plenty of good resources out there for that.
The only difference is when you install FreeSWITCH, you will want to compile with AMR Support, so that you can interact with mobile phones using the AMR codec, which I’ve documented how to do here.
Setting up your IMS
In order to get calls from the IMS to the Application Server, we need a way of routing the calls to the Application Server.
There are two standards-compliant ways to achieve this,
But this is a blunt instrument, after all, it’ll only ever be used at the start of the call, what if we want to send it to an AS because a destination can’t be reached and we want to play back a recorded announcement?
To support Dedicated Bearers we first have to have a way of profiling the traffic, to classify the traffic as being the type we want to provide the Dedicated Bearer for.
The first step involves a request from an Application Function (AF) to the PCRF via the Rx interface.
The most common type of AF would be a P-CSCF. When a VoLTE call gets setup the P-CSCF requests that a dedicated bearer be setup for the IP Address and Ports involved in the VoLTE call, to ensure users get the best possible call quality.
But Application Functions aren’t limited to just VoLTE – You could also embed an Application Function into the server for an online game to enable a dedicated bearer for users playing that game, or a sports streaming app that detects when a user starts streaming sports and creates a dedicated bearer for that user to send the traffic down.
The request to setup a dedicated bearer comes in the form of a Diameter request message from the AF, using the Rx reference point, typically from the P-CSCF to the PCRF in the network in an “AA-Request”.
Of main interest in the AA-Request is the Media Component AVP, that contains all the details needed to identify the traffic flow.
Now our PCRF is in charge of policy, and know which P-GW is serving the required subscriber. So the PCRF takes this information and sends a Gx Re-Auth Request to the PCEF in the P-GW serving the subscriber, with a Charging Rule the PCEF in the P-GW needs to install, to profile and apply QoS to the bearer.
Charging Rule Definition’s Flow-Information AVPs showing the information needed to profile the traffic
The QoS Description AVP defines which QoS parameters (QCI / ARP / Guaranteed & Maximum Bandwidth) should be applied to the traffic that matches the rules we just defined.
QoS Information AVP showing requested QoS Parameters
The P-GW sends back a Gx Re-Auth Answer, and gets to work actually setting up these bearers.
With the rule installed on the PCEF, it’s time to get this new bearer set up on the UE / eNodeB.
The P-GW sends a GTPv2 “Create Bearer Request” to the S-GW which forwards it onto the MME, to setup / define the Dedicated Bearer to be setup on the eNodeB.
GTPv2 “Create Bearer Request” sent by the P-Gw to the S-GW forwarded from the S-GW to the MME
The MME translates this into an S1 “E-RAB Setup Request” which it sends to the eNodeB to setup,
S1 E-RAB Setup request showing the E-RAB to be setup
Assuming the eNodeB has the resources to setup this bearer, it provides the details to the UE and sets up the bearer, sending confirmation back to the MME in the S1 “E-RAB Setup Response” message, which the MME translates back into GTPv2 for a “Create Bearer Response”
All this effort to keep your VoLTE calls sounding great!
Early on as subscriber trunk dialing and automated time-based charging was introduced to phone networks, engineers were faced with a problem from Payphones.
Previously a call had been a fixed price, once the caller put in their coins, if they put in enough coins, they could dial and stay on the line as long as they wanted.
But as the length of calls began to be metered, it means if I put $3 of coins into the payphone, and make a call to a destination that costs $1 per minute, then I should only be allowed to have a 3 minute long phone call, and the call should be cutoff before the 4th minute, as I would have used all my available credit.
Conversely if I put $3 into the Payphone and only call a $1 per minute destination for 2 minutes, I should get $1 refunded at the end of my call.
We see the exact same problem with prepaid subscribers on IMS Networks, and it’s solved in much the same way.
In LTE/EPC Networks, Diameter is used for all our credit control, with all online charging based on the Ro interface. So let’s take a look at how this works and what goes on.
Generic 3GPP Online Charging Architecture
3GPP defines a generic 3GPP Online charging architecture, that’s used by IMS for Credit Control of prepaid subscribers, but also for prepaid metering of data usage, other volume based flows, as well as event-based charging like SMS and MMS.
Network functions that handle chargeable services (like the data transferred through a P-GW or calls through a S-CSCF) contain a Charging Trigger Function (CTF) (While reading the specifications, you may be left thinking that the Charging Trigger Function is a separate entity, but more often than not, the CTF is built into the network element as an interface).
The CTF is a Diameter application that generates requests to the Online Charging Function (OCF) to be granted resources for the session / call / data flow, the subscriber wants to use, prior to granting them the service.
So network elements that need to charge for services in realtime contain a Charging Trigger Function (CTF) which in turn talks to an Online Charging Function (OCF) which typically is part of an Online Charging System (AKA OCS).
For example when a subscriber turns on their phone and a GTP session is setup on the P-GW/PCEF, but before data is allowed to flow through it, a Diameter “Credit Control Request” is generated by the Charging Trigger Function (CTF) in the P-GW/PCEF, which is sent to our Online Charging Server (OCS).
The “Credit Control Answer” back from the OCS indicates the subscriber has the balance needed to use data services, and specifies how much data up and down the subscriber has been granted to use.
The P-GW/PCEF grants service to the subscriber for the specified amount of units, and the subscriber can start using data.
This is a simplified example – Decentralized vs Centralized Rating and Unit Determination enter into this, session reservation, etc.
The interface between our Charging Trigger Functions (CTF) and the Online Charging Functions (OCF), is the Ro interface, which is a Diameter based interface, and is common not just for online charging for data usage, IMS Credit Control, MMS, value added services, etc.
3GPP define a reference online-charging interface, the Ro interface, and all the application-specific interfaces, like the Gy for billing data usage, build on top of the Ro interface spec.
Basic Credit Control Request / Credit Control Answer Process
This example will look at a VoLTE call over IMS.
When a subscriber sends an INVITE, the Charging Trigger Function baked in our S-CSCF sends a Diameter “Credit Control Request” (CCR) to our Online Charging Function, with the type INITIAL, meaning this is the first CCR for this session.
The CCR contains the Service Information AVP. It’s this little AVP that is where the majority of the magic happens, as it defines what the service the subscriber is requesting. The main difference between the multitude of online charging interfaces in EPC networks, is just what the service the customer is requesting, and the specifics of that service.
For this example it’s a voice call, so this Service Information AVP contains a “IMS-Information” AVP. This AVP defines all the parameters for a IMS phone call to be online charged, for a voice call, this is the called-party, calling party, SDP (for differentiating between voice / video, etc.).
It’s the contents of this Service Information AVP the OCS uses to make decision on if service should be granted or not, and how many service units to be granted. (If Centralized Rating and Unit Determination is used, we’ll cover that in another post) The actual logic, relating to this decision is typically based on the the rating and tariffing, credit control profiles, etc, and is outside the scope of the interface, but in short, the OCS will make a yes/no decision about if the subscriber should be granted access to the particular service, and if yes, then how many minutes / Bytes / Events should be granted.
In the received Credit Control Answer is received back from our OCS, and the Granted-Service-Unit AVP is analysed by the S-CSCF. For a voice call, the service units will be time. This tells the S-CSCF how long the call can go on before the S-CSCF will need to send another Credit Control Request, for the purposes of this example we’ll imagine the returned value is 600 seconds / 10 minutes.
The S-CSCF will then grant service, the subscriber can start their voice call, and start the countdown of the time granted by the OCS.
As our chatty subscriber stays on their call, the S-CSCF approaches the limit of the Granted Service units from the OCS (Say 500 seconds used of the 600 seconds granted). Before this limit is reached the S-CSCF’s CTF function sends another Credit Control Request with the type UPDATE_REQUEST. This allows the OCS to analyse the remaining balance of the subscriber and policies to tell the S-CSCF how long the call can continue to proceed for in the form of granted service units returned in the Credit Control Answer, which for our example can be 300 seconds.
Eventually, and before the second lot of granted units runs out, our subscriber ends the call, for a total talk time of 700 seconds.
But wait, the subscriber been granted 600 seconds for our INITIAL request, and a further 300 seconds in our UPDATE_REQUEST, for a total of 900 seconds, but the subscriber only used 700 seconds?
The S-CSCF sends a final Credit Control Request, this time with type TERMINATION_REQUEST and lets the OCS know via the Used-Service-Unit AVP, how many units the subscriber actually used (700 seconds), meaning the OCS will refund the balance for the gap of 200 seconds the subscriber didn’t use.
If this were the interface for online charging of data, we’d have the PS-Information AVP, or for online charging of SMS we’d have the SMS-Information, and so on.
The architecture and framework for how the charging works doesn’t change between a voice call, data traffic or messaging, just the particulars for the type of service we need to bill, as defined in the Service Information AVP, and the OCS making a decision on that based on if the subscriber should be granted service, and if yes, how many units of whatever type.
In the last we covered what ENUM is and how it works, so to take this into a more practical example, I thought I’d share the details of the ENUM server I’ve setup in my lab, and the Docker container I’ve bundled it into.
Inside the Docker container we’ll be running Bind – this post won’t teach you much about Bind, there’s already lots of good information on it elsewhere, but we will cover the parameters involved in setting up ENUM records (NAPTR) for E.164 addresses.
Getting the Environment up and Running
First we’ll need to setup our environment, I’ve published the images for the container to Dockerhub, but we’ll build it from the Dockerfile so you can edit the files and rebuild as you play around:
systemd-resolve on Ubuntu binds to port 53 by default, which can lead to some headaches, so we’ll create a new network in Docker for this to run in, so it doesn’t conflict with anything else you may be running:
And now we’ll run the ENUM container in the enum_playground network and with the IP 172.30.0.2,
docker run -d --rm --name=enum --net=enum_playground --ip=172.30.0.2 enum
Ok, that’s the environment setup, let’s run some queries!
E.164 to SIP URI Resolution with ENUM
In our last post we covered the basics of formatting an E.164 number and querying a DNS server to get it’s call routing information.
Again we’re going to use Dig to query this information. In reality ENUM queries would be run by an endpoint, or software like FreeSWITCH or Kamailio (Spoiler alert, posts on ENUM handling in those coming later), but as we’re just playing Dig will work fine.
So let’s start by querying a single E.164 address, +61355500911
First we’ll reverse it and put full stops / periods between the numbers, to get 1.1.9.0.0.5.5.5.3.1.6
Next we’ll add the e164.arpa prefix, which is the global prefix for ENUM addresses, and presto, that’s what we’ll query – 1.1.9.0.0.5.5.5.3.1.6.e164.arpa
Lastly we’ll feed this into a Dig query against the IP of our container and of type NAPTR,
Next up is the TTL or expiry, in this case it’s 3600 seconds (1 hour), shorter periods allow for changes to propagate / be reflected more quickly but at the expense of more load as results can’t be cached for as long. The class (IN) represents Internet, which is the only class commonly used, even on internal systems.
Then we have the type of record returned, in our case it’s a NAPTR record,
1.1.9.0.0.5.5.5.3.1.6.e164.arpa.3600 IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .
After that is the Order, this defines the order in which the rules are to be parsed. Lower numbers are processed first, if no matches then the next lowest, and so on until the highest number is reached, we’ll touch on this in more detail later in this post,
1.1.9.0.0.5.5.5.3.1.6.e164.arpa.3600 IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .
The Pref is the processing preference. This is very handy for load balancing, as we can split traffic between hosts with different preferences. We’ll cover this later in this post too.
1.1.9.0.0.5.5.5.3.1.6.e164.arpa.3600 IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .
The Flags represent the type of record we’re going to get, for most ENUM traffic this is going to be set to U, to denote a SIP URI with Regex, while the Service value we’ll be looking for will be “E2U+sip” service to identify SIP URIs to route calls to, but could be other values like Email addresses, IM Addresses or PSTN numbers, to be parsed by other applications.
1.1.9.0.0.5.5.5.3.1.6.e164.arpa.3600 IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .
Lastly we’ve got the Regex part. Again not going to cover Regex as a whole, just the DNS particulars.
Everything between the first and second ! denotes what we’re searching for, while everything from the second ! to the last ! denotes what we replace it with.
In the below example that means we’re matching ^.* which means starting with (^) any character (.) zero or more times (*), which gets replaced with sip:[email protected],
1.1.9.0.0.5.5.5.3.1.6.e164.arpa.3600 IN NAPTR 10 100"u" "E2U+sip" "!^.*$!sip:[email protected]!" .
How should this be treated?
For the first example, a call to the E.164 address of 61355500912 will be first formatted into a domain as per the ENUM requirements (1.1.9.0.0.5.5.5.3.1.6.e164.arpa) and then queried as a NAPTR record against the DNS server,
1.1.9.0.0.5.5.5.3.1.6.e164.arpa.3600 IN NAPTR 10 100"u" "E2U+sip" "!^.*$!sip:[email protected]!" .
Only a single record has been returned so we don’t need to worry about the Order or Preference, and the Regex matches anything and replaces it with the resulting SIP URI of sip:[email protected], which is where we’ll send our INVITE.
Under the Hood
Inside the Repo we cloned earlier, if you open the e164.arpa.db file, things will look somewhat familiar,
The record we just queried is the first example in the Bind config file,
; E.164 Address +61355500911 - Simple no replacement (Resolves all traffic to sip:[email protected])
1.1.9.0.0.5.5.5.3.1.6 IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .
The config file is just the domain, class, type, order, preference, flags, service and regex.
Astute readers may have noticed the trailing . which where we can put a replacement domain if Regex is not used, but it cannot be used in conjunction with Regex, so for all our work it’ll just be a single trailing . on each line.
You can (and probably should) change the values in the e164.arpa.db file as we go along to try everything out, you’ll just need to rebuild the container and restart it each time you make a change.
This post is going to focus on Bind, but the majority of modern DNS servers support NAPTR records, so you can use them for ENUM as well, for example I manage the DNS for this site thorough Cloudflare, and I’ve put a screenshot below of an example private ENUM address I’ve added into it.
Setting up a NAPTR record in Cloudflare DNS
Preference to Split Traffic between Servers
So with a firm understanding of a single record being returned, let’s look at how we can use ENUM to cleverly route traffic to multiple hosts.
If we have a pool of servers we may wish to evenly distribute all traffic across them, so that’s how E.164 address +61355500912 is setup – to route traffic evenly (50/50) across two servers.
Querying it with Dig provides the following result:
So as the order value (10) is the same for both records, we can ignore it – there isn’t one value lower than the other.
We can see both records have a preference of 100, in practice, this means they each get 50% of the traffic. The formula for traffic distribution is pretty simple, each server gets the value of it’s preference, divided by the total of all the preferences,
So for server1 it’s preference is 100 and the total of all the preferences combined is 200, so it gets 100/200, which is equivalent to one half aka 50%.
We might have a scenario where we have 3 servers, but one is significantly more powerful than the others, so let’s look at giving more traffic to one server and less to others, this example gets a little more complex but should cement your understanding of how the preference works;
So now 3 servers, again none have a lower order than the other, it’s set to 10 for them all so we can ignore the order,
Next we can see the total of all the priority values is 400,
Server 2 has a priority of 100 so it gets 100/400 total priority, or a quarter of all traffic. Server 1 has the same value, so also gets a quarter of all traffic,
Server 3 however has a priority of 200 so it gets 200/400, or to simplify half of all traffic.
The Bind config for this is:
; E.164 Address +61355500913 - More complex load balance between 3 hosts (25% server1, 25% server2, 50% server3)
3.1.9.0.0.5.5.5.3.1.6 IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" . 3.1.9.0.0.5.5.5.3.1.6 IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .
3.1.9.0.0.5.5.5.3.1.6 IN NAPTR 10 200 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .
Order for Failover
Primarily the purpose of the order is to enable wildcard routes (as we’ll see later) to be overwritten by more specific routes, but a secondary use in some implementations use Order as a way to list the preferences of the SIP URIs to route to. For example we could have two servers, one a primary and the other a standby, with the standby only to be used only if the primary SIP URI was not responding.
E.164 number +61355500914 is setup to return two SIP URIs,
Our DNS client will first use the SIP URI sip:[email protected] as it has the lower order value (10), and if that fails, can try the entry with the next lowest order-value (20) which would be sip:[email protected].
The Bind config for this is:
; E.164 Address +61355500914 - Order example returning multiple SIP URIs to try for failover
4.1.9.0.0.5.5.5.3.1.6 IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" . 4.1.9.0.0.5.5.5.3.1.6 IN NAPTR 20 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .
Wildcards
If we have a 1,000 number block, having to add 1000 individual records can be very tedious. Instead we can use wildcard matching (thanks to the fact we’ve reversed the E.164 address) to match ranges. For example if we have E.164 numbers from +61255501000 to +61255501999 we can add a wildcard entry to match the +61255501x prefix,
I’ve set this up already so let’s lookup the E.164 number +6125501234,
If you look up any other number starting with +6125501 you’ll get the same result, and here’s the Bind config for it:
; Wildcard E.164 Address +61255501* - Wildcard example for all destinations starting with E.164 prefix +61255501x to single destination (sip:[email protected])
; For example E.164 number +6125501234 will resolve to sip:[email protected]
*.1.0.5.5.5.2.1.6 IN NAPTR 100 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .
The catch with this is they’re all pointing at the same SIP URI, so we can’t treat the calls differently based on the called number – This is where the Regex magic comes in.
We can use group matching to match a group and fill it in the dialed number into the SIP Request URI, for example:
Will match the E.164 number requested and put it inside sip:[email protected]
The +61255502xxx prefix is setup for this, so if we query +61255502000 (or any other number between +61255502000 and +61255502999) we’ll get the regex query in the resulting record.
Keep in mind DNS doesn’t actually apply the Regex transformation, just shares it, and the client applies the transformation.
; Wildcard example for all destinations starting with E.164 prefix +61255502x to regex filled destination
; For example a request to 61255502000 will return sip:[email protected])
*.2.0.5.5.5.2.1.6 IN NAPTR 100 100 "u" "E2U+sip" "!(^.*$)!sip:+1\\[email protected]!" .
One last thing to keep in mind, is that Wildcard priorities are of any length. This means +612555021 would match as well as +6125550299999999999999. Typically terminating switches drop any superfluous digits, and NU those that are too short, but keep this in mind, that length is not taken into account.
Wildcard Priorities
So with our wildcards in place, what if we wanted to add an exception, for example one number in our 61255502xxx block of numbers gets ported to another carrier and needs to be routed elsewhere?
Easy, we just add another entry for that number being more specific and with a lower order than the wildcard, which is what’s setup for E.164 number +61255502345,
Which does not return the same result as the others that match the wildcard,
Bind config:
; Wildcard example for all destinations starting with E.164 prefix +61255502x to regex filled destination
; For example a request to +61255502000 will return sip:[email protected])
*.2.0.5.5.5.2.1.6 IN NAPTR 100 100 "u" "E2U+sip" "!(^.*$)!sip:+1\\[email protected]!" .
; More specific example with lower order than +6125550x wildcard for E.164 address +61255502345 will return sip:[email protected]
5.4.3.2.0.5.5.5.2.1.6 IN NAPTR 50 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .
We can combine all of the tricks we’ve covered here, from statically defined entries, wildcards, regex replacement, multiple entries with multiple orders and preferences, to create really complex routing, using only DNS.
Summary & Next Steps
So by now hopefully you’ve got a fair understanding of how NAPTR and DNS work together to translate E.164 addresses into SIP URIs,
Of course being able to do this manually with Dig and comprehend how it’ll route is only one part of the picture, in the next posts we’ll cover using Kamailio and FreeSWITCH to query ENUM routing information and route traffic to it,
SIP routing is complicated, there’s edge cases, traffic that can be switched locally and other traffic that needs to be proxied off to another Proxy or Application server. How can you define these rules and logic in a flexible way, that allows these rules to be distributed out to multiple different network elements and adjusted on a per-subscriber basis?
Enter iFCs – The Initial Filter Criteria.
iFCs are XML encoded rules to define which servers should handle traffic matching a set of rules.
Let’s look at some example rules we might want to handle through iFCs:
Send all SIP NOTIFY, SUBSCRIBE and PUBLISH requests to a presence server
Any Mobile Originated SMS to an SMSc
Calls to a specific destination to a MGC
Route any SIP INVITE requests with video codecs present to a VC bridge
Send calls to Subscribers who aren’t registered to a Voicemail server
Use 3rd party registration to alert a server that a Subscriber has registered
All of these can be defined and executed through iFCs, so let’s take a look,
iFC Structure
iFCs are encoded in XML and typically contained in the Cx-user-data AVP presented in a Cx Server Assignment Answer response.
Let’s take a look at an example iFC and then break down the details as to what we’re specifying.
Each rule in an iFC is made up of a Priority, TriggerPoint and ApplicationServer.
So for starters we’ll look at the Priority tag. The Priority tag allows us to have multiple-tiers of priority and multiple levels of matching, For example if we had traffic matching the conditions outlined in this rule (TriggerPoint) but also matching another rule with a lower priority, the lower priority rule would take precedence.
Inside our <TriggerPoint> tag contains the specifics of the rules and how the rules will be joined / matched, which is what we’ll focus on predominantly, and is followed by the <ApplicationServer> which is where we will route the traffic to if the TriggerPoint is matched / triggered.
So let’s look a bit more about what’s going on inside the TriggerPoint.
Each TriggerPoint is made up of Service Point Trigger (SPTs) which are individual rules that are matched or not matched, that are either combined as logical AND or logical OR statements when evaluated.
By using fairly simple building blocks of SPTs we can create a complex set of rules by joining them together.
Service Point Triggers (SPTs)
Let’s take a closer look at what goes on in an SPT. Below is a simple SPT that will match all SIP requests using the SIP MESSAGE method request type:
So as you may have guessed, the <Method> tag inside the SPT defines what SIP request method we’re going to match.
But Method is only one example of the matching mechanism we can use, but we can also match on other attributes, such as Request URI, SIP Header, Session Case (Mobile Originated vs Mobile Terminated) and Session Description such as SDP.
Or an example of a SPT for anything Originating from the Subscriber utilizing the <SessionCase> tag inside the SPT.
Having <Header> will match if the header is present, while the optional Content tag can be used to match
In terms of the Content this is matched using Regular Expressions, but in this case, not so regular regular expressions. 3GPP selected Extended Regular Expressions (ERE) to be used (IEEE POSIX) which are similar to the de facto standard PCRE Regex, but with a few fewer parameters.
Condition Negated
The <ConditionNegated> tag inside the SPT allows us to do an inverse match.
In short it will match anything other than what is specified in the SPT.
For example if we wanted to match any SIP Methods other than MESSAGE, setting <ConditionNegated>1</ConditionNegated> would do just that, as shown below:
Finally the <Group> tag allows us to group together a group of rules for the purpose of evaluating. We’ll go into it more in in the below section.
ConditionTypeCNF / ConditionTypeDNF
As we touched on earlier, <TriggerPoints> contain all the SPTs, but also, very importantly, specify how they will be interpreted.
SPTs can be joined in AND or OR conditions.
For some scenarios we may want to match where METHOD is MESSAGE and RequestURI is sip:[email protected], which is different to matching where the METHOD is MESSAGE or RequestURI is sip:[email protected].
This behaviour is set by the presence of one of the ConditionTypeCNF (Conjunctive Normal Form) or ConditionTypeDNF (Disjunctive Normal Form) tags.
If each SPT has a unique number in the GroupTag and ConditionTypeCNF is set then we evaluate as AND.
If each SPT has a unique number in the GroupTag and ConditionTypeDNF is set then we evaluate as OR.
Let’s look at how the below rule is evaluated as AND as ConditionTypeCNF is set:
This means we will match if the method is MESSAGE and Session Case is 0 (Mobile Originated) as each SPT is in a different Group which leads to “and” behaviour.
If we were to flip to ConditionTypeDNF each of the SPTs are evaluated as OR.
If you’re building IMS Networks, the AMR config is a must, but FreeSWITCH does not ship with AMR due to licencing constraints, but has all the hard work done, you just need to add the headers for AMR support and compile.
LibOpenCore has support for AMR which we build, and then with a few minor tweaks to copy the C++ header files over to the FreeSWITCH source directory, and enable support in modules.conf.
Then when building FreeSWITCH you’ve got the AMR Codec to enable you to manage IMS / VoLTE media streams from mobile devices.
Instead of copying and pasting a list of commands to do this, I’ve published a Dockerfile here you can use to build a Docker image, or on a straight Debian Buster machine if you’re working on VMs or Bare Metal if you run the commands from the Dockerfile on the VM / bare metal.
Since the beginning of time, SIP has used the 2xx responses to confirm all went OK.
If you thought sending an SMS in a VoLTE/IMS network would see a 2xx OK response and then that’s the end of it, you’d be wrong.
So let’s take a look into sending SMS over VoLTE/IMS networks!
So our story starts with the Subscriber sending an SMS, which generate a SIP MESSAGE.
The Content-Type of this SIP MESSAGE is set to application/vnd.3gpp.sms rather than Text, and that’s because SMS over IMS uses the Short Message Transfer Protocol (SM-TP) inherited from GSM.
The Short Message Transfer Protocol (SM-TP) (Not related to Simple Message Transfer Protocol used in Email clients) is made up of Transfer Protocol Data Units (TPDU) that contain our message information, even though we have the Destination in our SIP headers, it’s again defined in the SM-TP body.
At first this may seem like a bit of duplication, but this allows older SMS Switching Centers (SMSc) to add support for IMS networks without any major changes, just what the SM-TP payload is wrapped up in changes.
SIP MESSAGE Request Body encoded in SM-TP
So back to our SIP MESSAGE request, typed out by the Subscriber, the UE sends this a SIP MESSAGE onto our IMS Network.
The IMS network follows it’s IFCs and routing rules, and makes it to the termination points for SMS traffic – the SMSc.
The SMSc sends back either a 200 OK or a 202 Accepted, and you’d think that’s the end of it, but no.
Our Subscriber still sees “Sending” on the screen, and the SMS is not shown as sent yet.
Instead, when the SMS has been delivered or buffered, relayed, etc, the SMSc generates a new SIP request, (as in new Call-ID / Dialog) with the request type MESSAGE, addressed to the Subscriber.
The payload of this request is another application/vnd.3gpp.sms encoded request body, again, containing SM-TP encoded data.
When the UE receives this, it will then consider the message delivered.
SM-TP encoded Delivery Report
Of course things change slightly when delivery reports are enabled, but that’s another story!
These posts focus on the use of Diameter and SIP in an IMS / VoLTE context, however these practices can be equally applied to other networks.
The Registration-Termination Request / Answer allow a Diameter Client (S-CSCF) to indicate to the HSS (Diameter Server) that it is no longer serving that user and the registration has been terminated.
Basics:
The RFC’s definition is actually pretty succinct as to the function of the Server-Assignment Request/Answer:
The Registration-Termination-Request is sent by a Diameter Multimedia server to a Diameter Multimedia client in order to request the de-registration of a user.
Reference: TS 29.229
The Registration-Termination-Request commands are sent by a S-CSCF to indicate to the Diameter server that it is no longer serving a specific subscriber, and therefore this subscriber is now unregistered.
There are a variety of reasons for this, such as PERMANENT_TERMINATION, NEW_SIP_SERVER_ASSIGNED and SIP_SERVER_CHANGE.
The Diameter Server (HSS) will typically send the Diameter Client (S-CSCF) a Registration-Termination-Answer in response to indicate it has updated it’s internal database and will no longer consider the user to be registered at that S-CSCF.
Packet Capture
I’ve included a packet capture of these Diameter Commands from my lab network which you can find below.
These posts focus on the use of Diameter and SIP in an IMS / VoLTE context, however these practices can be equally applied to other networks.
The Diameter User-Authorization-Request and User-Authorization-Answer commands are used as the first line of authorization of a user and to determine which Serving-CSCF to forward a request to.
Basics
When a SIP Proxy (I-CSCF) receives an incoming SIP REGISTER request, it sends a User-Authorization-Request to a Diameter server to confirm if the user exists on the network, and which S-CSCF to forward the request to.
When the Diameter server receives the User-Authorization-Request it looks at the User-Name (1) AVP to determine if the Domain / Realm is served by the Diameter server and the User specified exists.
Assuming the user & domain are valid, the Diameter server sends back a User-Authorization-Answer, containing a Server-Capabilities (603) AVP with the Server-Name of the S-CSCF the user will be served by.
I always find looking at the packets puts everything in context, so here’s a packet capture of both the User-Authorization-Request and the User-Authorization-Answer.
Wireshark display of User-Authorization-Request packet
Wireshark display of User-Authorization-Answer packet
First Registration
If this is the first time this Username / Domain combination (Referred to in the RFC as an AOR – Address of Record) is seen by the Diameter server in the User-Authorization-Request it will allocate a S-CSCF address for the subscriber to use from it’s pool / internal logic.
The Diameter server will store the S-CSCF it allocated to that Username / Domain combination (AoR) for subsequent requests to ensure they’re routed to the same S-CSCF.
The Diameter server indicates this is the first time it’s seen it by adding the DIAMETER_FIRST_REGISTRATION (2001) AVP to the User-Authorization-Answer.
Subsequent Registration
If the Diameter server receives another User-Authorization-Request for the same Username / Domain (AoR) it has served before, the Diameter server returns the same S-CSCF address as it did in the first User-Authorization-Answer.
It indicates this is a subsequent registration in much the same way the first registration is indicated, by adding an DIAMETER_SUBSEQUENT_REGISTRATION (2002) AVP to the User-Authorization-Answer.
User-Authorization-Type (623) AVP
An optional User-Authorization-Type (623) AVP is available to indicate the reason for the User-Authorization-Request. The possible values / reasons are:
Creating / Updating / Renewing a SIP Registration (REGISTRATION (0))
Establishing Server Capabilities & Registering (CAPABILITIES (2))
Terminating a SIP Registration (DEREGISTRATION (1))
If the User-Authorization-Type is set to DEREGISTRATION (1) then the Diameter server returns the S-CSCF address in the User-Authorization-Answer and then removes the S-SCSF address it had associated with the AoR from it’s own records.
These posts focus on the use of Diameter and SIP in an IMS / VoLTE context, however these practices can be equally applied to other networks.
The Server-Assignment-Request/Answer commands are used so a SIP Server can indicate to a Diameter server that it is serving a subscriber and pull the profile information of the subscriber.
Basics:
The RFC’s definition is actually pretty succinct as to the function of the Server-Assignment Request/Answer:
The main functions of the Diameter SAR command are to inform the Diameter server of the URI of the SIP server allocated to the user, and to store or clear it from the Diameter server.
Additionally, the Diameter client can request to download the user profile or part of it.
The Server-Assignment-Request/Answer commands are sent by a S-CSCF to indicate to the Diameter server that it is now serving a specific subscriber, (This information can then be queried using the Location-Info-Request commands) and get the subscriber’s profile, which contains the details and identities of the subscriber.
Typically upon completion of a successful SIP REGISTER dialog (Multimedia-Authentication Request), the SIP Server (S-CSCF) sends the Diameter server a Server-Assignment-Request containing the SIP Username / Domain (referred to as an Address on Record (SIP-AOR) in the RFC) and the SIP Server (S-CSCF)’s SIP-Server-URI.
The Diameter server looks at the SIP-AOR and ensures there are not currently any active SIP-Server-URIs associated with that AoR. If there are not any currently active it then stores the SIP-AOR and the SIP-Server-URI of the SIP Server (S-CSCF) serving that user & sends back a Server-Assignment-Answer.
For most request the Subscriber’s profile is also transfered to the S-SCSF in the Server-Assignment-Answer command.
SIP-Server-Assignment-Type AVP
The same Server-Assignment-Request command can be used to register, re-register, remove registration bindings and pull the user profile, through the information in the SIP-Server-Assignment-Type AVP (375),
Common values are:
NO_ASSIGNMENT (0) – Used to pull just the user profile
The Cx-User-Data profile contains the subscriber’s profile from the Diameter server in an XML formatted dataset, that is contained as part of the Server-Assignment-Answer in the Cx-User-Data AVP (606).
The profile his tells the S-CSCF what services are offered to the subscriber, such as the allowed SIP Methods (ie INVITE, MESSAGE, etc), and how to handle calls to the user when the user is not registered (ie send calls to voicemail if the user is not there).
There’s a lot to cover on the user profile which we’ll touch on in a later post.
These posts focus on the use of Diameter and SIP in an IMS / VoLTE context, however these practices can be equally applied to other networks.
The Location-Information-Request/Answer commands are used so a SIP Server query a Diameter to find which P-CSCF a Subscriber is being served by
Basics:
The RFC’s definition is actually pretty succinct as to the function of the Server-Assignment Request/Answer:
The Location-Info-Request is sent by a Diameter Multimedia client to a Diameter Multimedia server in order to request name of the server that is currently serving the user.Reference: 29.229-
The Location-Info-Request is sent by a Diameter Multimedia client to a Diameter Multimedia server in order to request name of the server that is currently serving the user.
Reference: TS 29.229
The Location-Info-Request commands is sent by an I-CSCF to the HSS to find out from the Diameter server the FQDN of the S-CSCF serving that user.
The Public-Identity AVP (601) contains the Public Identity of the user being sought.
Here you can see the I-CSCF querying the HSS via Diameter to find the S-CSCF for public identity 12722123
The Diameter server sends back the Location-Info-Response containing the Server-Name AVP (602) with the FQDN of the S-CSCF.
Packet Capture
I’ve included a packet capture of these Diameter Commands from my lab network which you can find below.
These posts focus on the use of Diameter and SIP in an IMS / VoLTE context, however these practices can be equally applied to other networks.
The Multimedia-Authentication-Request/Answer commands are used to Authenticate subscribers / UAs using a variety of mechanisms such as straight MD5 and AKAv1-MD5.
Basics:
When a SIP Server (S-CSCF) receives a SIP INVITE, SIP REGISTER or any other SIP request, it needs a way to Authenticate the Subscriber / UA who sent the request.
We’ve already looked at the Diameter User-Authorization-Request/Answer commands used to Authorize a user for access, but the Multimedia-Authentication-Request / Multimedia-Authentication-Answer it used to authenticate the user.
The SIP Server (S-CSCF) sends a Multimedia-Authentication-Request to the Diameter server, containing the Username of the user attempting to authenticate and their Public Identity.
The Diameter server generates “Authentication Vectors” – these are Precomputed cryptographic challenges to challenge the user, and the correct (“expected”) responses to the challenges. The Diameter puts these Authentication Vectors in the 3GPP-SIP-Auth-Data (612) AVP, and sends them back to the SIP server in the Multimedia-Authentication-Answer command.
The SIP server sends the Subscriber / UA a SIP 401 Unauthorized response to the initial request, containing a WWW-Authenticate header containing the challenges.
SIP 401 Response with WWW-Authenticate header populated with values from Multimedia-Auth-Answer
The Subscriber / UA sends back the initial request with the WWW-Authenticate header populated to include a response to the challenges. If the response to the challenge matches the correct (“expected”) response, then the user is authenticated.
Multimedia-Authentication-Request
Multimedia-Authentication-Answer
I always find it much easier to understand what’s going on through a packet capture, so here’s a packet capture showing the two Diameter commands,
Note: There is a variant of this process allows for stateless proxies to handle this by not storing the expected authentication values sent by the Diameter server on the SIP Proxy, but instead sending the received authentication values sent by the Subscriber/UA to the Diameter server to compare against the expected / correct values.
The Cryptography
The Cryptography for IMS Authentication relies on AKAv1-MD5 which I’ve written about before,
Essentially it’s mutual network authentication, meaning the network authenticates the subscriber, but the subscriber also authenticates the network.
I had a few headaches getting the example P-CSCF example configs from the Kamailio team to run, recent improvements with the IPsec support and code evolution meant that the example config just didn’t run.
So, after finally working out the changes I needed to make to get Kamailio to function as a P-CSCF, I took the plunge and made my first pull request on the Kamailio project.
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.
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.
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.
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.
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.
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 Security–Server 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.
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
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.
While poking around the development and debugging features on Samsung handsets I found the ability to run IMS Debugging directly from the handset.
Alas, the option is only available in the commercial version, it’s just there for carriers, and requires a One Time Password to unlock.
When tapping on the option a challenge is generated with a key.
Interestingly I noticed that the key changes each time and can reject you even in aeroplane mode, suggesting the authentication happens client side.
This left me thinking – If the authentication happens client side, then the App has to know what the valid password for the key shown is…
Some research revealed you can pull APKs off an Android phone, so I downloaded a utility called “APK Extractor” from the Play store, and used it to extract the Samsung Sysdump utility.
So now I was armed with the APK on my local machine, the next step was to see if I could decompile the APK back into source code.
Some Googling found me an online APK decompiler, which I fed the compiled APK file and got back the source code.
I did some poking around inside the source code, and then I found an interesting directory:
Here’s a screenshot of the vanilla code that came out of the app.
I’m not a Java expert, but even I could see the “CheckOTP” function and understand that that’s what validates the One Time Passwords.
The while loop threw me a little – until I read through the rest of the code; the “key” in the popup box is actually a text string representing the current UNIX timestamp down to the minute level. The correct password is an operation done on the “key”, however the CheckOTP function doesn’t know the challenge key, but has the current time, so generates a challenge key for each timestamp back a few minutes and a few minutes into the future.
I modified the code slightly to allow me to enter the presented “key” and get the correct password back. It’s worth noting you need to act quickly, enter the “key” and enter the response within a minute or so.
In the end I’ve posted the code on an online Java compiler,
