Tag Archives: IMS

Improving WiFi Calling quality for WiFi Operators

I had a question recently on LinkedIn regarding how to preference Voice over WiFi traffic so that a network engineer operating the WiFi network can ensure the best quality of experience for Voice over WiFi.

Voice over WiFi is underpinned by the ePDG – Evolved Packet Data Gateway (this is a fancy IPsec tunnel we authenticate to using the SIM to drop our traffic into the P-CSCF over an unsecured connection). To someone operating a WiFi network, the question is how do we prioritise the traffic to the ePDGs and profile it?

ePDGs can be easily discovered through a simple DNS lookup, once you know the Mobile Network Code and Mobile Country code of the operators you want to prioritise, you can find the IPs really easily.

ePDG addresses take the form epdg.epc.mncXXX.mccYYY.pub.3gppnetwork.org so let’s look at finding the IPs for each of these for the operators in a country:

The first step is nailing down the mobile network code and mobile country codes of the operators you want to target, Wikipedia is a great source for this information.
Here in Australia we have the Mobile Country Code 505 and the big 3 operators all support Voice over WiFi, so let’s look at how we’d find the IPs for each.
Telstra has mobile network code (MNC) 01, in 3GPP DNS we always pad network codes to 3 digits, so that’ll be 001, and the mobile country code (MCC) for Australia is 505.
So to find the IPs for Telstra we’d run an nslookup for epdg.epc.mnc001.mcc505.pub.3gppnetwork.org – The list of IPs that are returned, are the IPs you’ll see Voice over WiFi traffic going to, and the IPs you should provide higher priority to:

For the other big operators in Australia epdg.epc.mnc002.mcc505.pub.3gppnetwork.org will get you Optus and epdg.epc.mnc003.mcc505.pub.3gppnetwork.org will get you VHA.

The same rules apply in other countries, you’d just need to update the MNC/MCC to match the operators in your country, do an nslookup and prioritise those IPs.

Generally these IPs are pretty static, but there will need to be a certain level of maintenance required to keep this list up to date by rechecking.

Happy WiFi Calling!

Verify Android Signing Certificate for ARA-M Carrier Privileges in App

Part of the headache when adding the ARA-M Certificate to a SIM is getting the correct certificate loaded,

The below command calculates it the SHA-1 Digest we need to load as the App ID on the SIM card’s ARA-M or ARA-F applet:

apksigner verify --verbose --print-certs "yourapp.apk"

You can then flash this onto the SIM with PySIM:

pySIM-shell (MF/ADF.ARA-M)> aram_store_ref_ar_do --aid FFFFFFFFFFFF --device-app-id E46872F28B350B7E1F140DE535C2A8D5804F0BE3 --android-permissions 0000000000000001 --apdu-always

Authenticating Fixed Line Subscribers into IMS

We recently added support in PyHSS for fixed line SIP subscribers to attach to the IMS.

Traditional telecom operators are finding their fixed line network to be a bit of a money pit, something they’re required to keep operating to meet regulatory obligations, but the switches are sitting idle 99% of the time. As such we’re seeing more and more operators move fixed line subs onto their IMS.

This new feature means we can use PyHSS to serve as the brains for a fixed network, as well as for mobile, but there’s one catch – How we authenticate subscribers changes.

Most banks of line cards in a legacy telecom switches, or IP Phones, don’t have SIM slots to allow us to authenticate, so instead we’re forced to fallback to what they do support.

Unfortunately for the most part, what is supported by these IP phones or telecom switches is SIP MD5 Digest Authentication.

The Nonce is generated by the HSS and put into the Multimedia-Authentication-Answer, along with the subscriber’s password and sent in the clear to the S-CSCF.

Subscriber with Password made up of all 1's MAA response from HSS for Digest-MD5 Auth

The HSS then generates the the Multimedia-Auth Answer, it generates a nonce (in the 3GPP-SIP-Authenticate / 609 AVP) and sends the Subscriber’s password in the 3GPP-SIP-Authorization (610) AVP in response back to the S-CSCF.

I would have thought a better option would be for the HSS to generate the Nonce and Digest, and then the S-CSCF to just send the Nonce to the Sub and compare the returned Digest from the Sub against the expected Digest from the HSS, but it would limit flexibility (realm adaptation, etc) I guess.

The UE/UA (I guess it’s a UA in this context as it’s not a mobile) then generates its own Digest from the Nonce and sends it back to the S-CSCF via the P-CSCF.

The S-CSCF compares the received Digest response against the one it generated, and if the two match, the sub is authenticated and allowed to attach onto the network.

IMS iFC – SPT Session Cases

Mostly just reference material for me:

Possible values:

  • 0 (ORIGINATING_SESSION)
  • 1 TERMINATING_REGISTERED
  • 2 (TERMINATING_UNREGISTERED)
  • 3 (ORIGINATING_UNREGISTERED

In the past I had my iFCs setup to look for the P-Access-Network-Info header to know if the call was coming from the IMS, but it wasn’t foolproof – Fixed line IMS subs didn’t have this header.

            <TriggerPoint>
                <ConditionTypeCNF>1</ConditionTypeCNF>
                <SPT>
                    <ConditionNegated>0</ConditionNegated>
                    <Group>0</Group>
                    <Method>INVITE</Method>
                    <Extension></Extension>
                </SPT>
                <SPT>
                    <ConditionNegated>0</ConditionNegated>
                    <Group>1</Group>
                    <SIPHeader>
                      <Header>P-Access-Network-Info</Header>
                    </SIPHeader>
                </SPT>                
            </TriggerPoint>

But now I’m using the Session Cases to know if the call is coming from a registered IMS user:

        <!-- SIP INVITE Traffic from Registered Sub-->
        <InitialFilterCriteria>
            <Priority>30</Priority>
            <TriggerPoint>
                <ConditionTypeCNF>1</ConditionTypeCNF>
                <SPT>
                    <ConditionNegated>0</ConditionNegated>
                    <Group>0</Group>
                    <Method>INVITE</Method>
                    <Extension></Extension>
                </SPT>
                <SPT>
                    <Group>0</Group>
                    <SessionCase>0</SessionCase>
                </SPT>             
            </TriggerPoint>

SQN Sync in IMS Auth

So the issue was a head scratcher.

Everything was working on the IMS, then I go to bed, the next morning I fire up the test device and it just won’t authenticate to the IMS – The S-CSCF generated a 401 in response to the REGISTER, but the next REGISTER wouldn’t pass.

Wireshark just shows me this loop:

UE -> IMS: REGISTER
IMS -> UE: 401 Unauthorized (With Challenge)
UE -> IMS: REGISTER with response
IMS -> UE: 401 Unauthorized (With Challenge)
UE -> IMS: REGISTER with response
IMS -> UE: 401 Unauthorized (With Challenge)
UE -> IMS: REGISTER with response
IMS -> UE: 401 Unauthorized (With Challenge)

So what’s going on here?

IMS uses AKAv1-MD5 for Authentication, this is slightly different to the standard AKA auth used in cellular, but if you’re curious, we’ve covered by IMS Authentication and standard AKA based SIM Authentication in cellular networks before.

When we generate the vectors (for IMS auth and standard auth) one of the inputs to generate the vectors is the Sequence Number or SQN.

This SQN ticks over like an odometer for the number of times the SIM / HSS authentication process has been performed.

There is some leeway in the SQN – It may not always match between the SIM and the HSS and that’s to be expected.
When the MME sends an Authentication-Information-Request it can ask for multiple vectors so it’s got some in reserve for the next time the subscriber attaches, and that’s allowed.

Information stored on USIM / SIM Card for LTE / EUTRAN / EPC - K key, OP/OPc key and SQN Sequence Number

But there are limits to how far out our SQN can be, and for good reason – One of the key purposes for the SQN is to protect against replay attacks, where the same vector is replayed to the UE. So the SQN on the HSS can be ahead of the SIM (within reason), but it can’t be behind – Odometers don’t go backwards.

So the issue was with the SQN on the SIM being out of Sync with the SQN in the IMS, how do we know this is the case, and how do we fix this?

Well there is a resync mechanism so the SIM can securely tell the HSS what the current SQN it is using, so the HSS can update it’s SQN.

When verifying the AUTN, the client may detect that the sequence numbers between the client and the server have fallen out of sync.
In this case, the client produces a synchronization parameter AUTS, using the shared secret K and the client sequence number SQN.
The AUTS parameter is delivered to the network in the authentication response, and the authentication can be tried again based on authentication vectors generated with the synchronized sequence number.

RFC 3110: HTTP Digest Authentication using AKA

In our example we can tell the sub is out of sync as in our Multimedia Authentication Request we see the SIP-Authorization AVP, which contains the AUTS (client synchronization parameter) which the SIM generated and the UE sent back to the S-CSCF. Our HSS can use the AUTS value to determine the correct SQN.

SIP-Authorization AVP in the Multimedia Authentication Request means the SQN is out of Sync and this AVP contains the RAND and AUTN required to Resync

Note: The SIP-Authorization AVP actually contains both the RAND and the AUTN concatenated together, so in the above example the first 32 bytes are the AUTN value, and the last 32 bytes are the RAND value.

So the HSS gets the AUTS and from it is able to calculate the correct SQN to use.

Then the HSS just generates a new Multimedia Authentication Answer with a new vector using the correct SQN, sends it back to the IMS and presto, the UE can respond to the challenge normally.

This feature is now fully implemented in PyHSS for anyone wanting to have a play with it and see how it all works.

And that friends, is how we do SQN resync in IMS!

Failures in cobbling together a USSD Gateway

One day recently I was messing with the XCAP server, trying to set the Call Forward timeout. In the process I triggered the UE to send a USSD request to the IMS.

Huh, I thought, “I wonder how hard it would be to build a USSD Gateway for our IMS?”, and this my friends, is the story of how I wasted a good chunk of my weekend trying (and failing) to add support for USSD.

You might be asking “Who still uses USSD?” – The use cases for USSD are pretty thin on the ground in this day and age, but I guess balance query, and uh…

But this is the story of what I tried before giving up and going outside…

Routing

First I’d need to get the USSD traffic towards the USSD Gateway, this means modifying iFCs. Skimming over the spec I can see the Recv-Info: header for USSD traffic should be set to “g.3gpp.ussd” so I knocked up an iFC to match that, and route the traffic to my dev USSD Gateway, and added it to the subscriber profile in PyHSS:

  <!-- SIP USSD Traffic to USSD-GW-->
        <InitialFilterCriteria>
            <Priority>25</Priority>
            <TriggerPoint>
                <ConditionTypeCNF>1</ConditionTypeCNF>
                <SPT>
                    <ConditionNegated>0</ConditionNegated>
                    <Group>1</Group>
                    <SIPHeader>
                      <Header>Recv-Info</Header>
                      <Content>"g.3gpp.ussd"</Content>
                    </SIPHeader>
                </SPT>                
            </TriggerPoint>
            <ApplicationServer>
                <ServerName>sip:ussdgw:5060</ServerName>
                <DefaultHandling>0</DefaultHandling>
            </ApplicationServer>
        </InitialFilterCriteria>

Easy peasy, now we have the USSD requests hitting our USSD Gateway.

The Response

I’ll admit that I didn’t jump straight to the TS doc from the start.

The first place I headed was Google to see if I could find any PCAPs of USSD over IMS/SIP.

And I did – Restcomm seems to have had a USSD product a few years back, and trawling around their stuff provided some reference PCAPs of USSD over SIP.

So the flow seemed pretty simple, SIP INVITE to set up the session, SIP INFO for in-dialog responses and a BYE at the end.

With all the USSD guts transferred as XML bodies, in a way that’s pretty easy to understand.

Being a Kamailio fan, that’s the first place I started, but quickly realised that SIP proxies, aren’t great at acting as the UAS.

So I needed to generate in-dialog SIP INFO messages, so I turned to the UAC module to generate the SIP INFO response.

My Kamailio code is super simple, but let’s have a look:

request_route {

        xlog("Request $rm from $fU");

        if(is_method("INVITE")){
                xlog("USSD from $fU to $rU (Emergency number) CSeq is $cs ");
                sl_reply("200", "OK Trying USSD Phase 1");      #Generate 200 OK
                route("USSD_Response"); #Call USSD_Response route block
                exit;
        }
}

route["USSD_Response"]{
        xlog("USSD_Response Route");
        #Generate a new UAC Request
        $uac_req(method)="INFO";
        $uac_req(ruri)=$fu;     #Copy From URI to Request URI
        $uac_req(furi)=$tu;     #Copy To URI to From URI
        $uac_req(turi)=$fu;     #Copy From URI to To URI
        $uac_req(callid)=$ci;   #Copy Call-ID
                                #Set Content Type to 3GPP USSD
        $uac_req(hdrs)=$uac_req(hdrs) + "Content-Type: application/vnd.3gpp.ussd+xml\r\n";
                                #Set the USSD XML Response body
        $uac_req(body)="<?xml version='1.0' encoding='UTF-8'?>
        <ussd-data>
                <language value=\"en\"/>
                <ussd-string value=\"Bienvenido. Seleccione una opcion: 1 o 2.\"/>
        </ussd-data>";
        $uac_req(evroute)=1;    #Set the event route to use on return replies
        uac_req_send();         #Send it!
}

So the UAC module generates the 200 OK and sends it back.

“That was quick” I told myself, patting myself on the back before trying it out for the first time.

Huston, we have a problem – Although the Call-ID is the same, it’s not an in-dialog response as the tags aren’t present, this means our UE send back a 405 to the SIP INFO.

Right. Perhaps this is the time to read the Spec…

Okay, so the SIP INFO needs to be in dialog. Can we do that with the UAC module? Perhaps not…

But the Transaction Module ™ in Kamailio exposes and option on the ctl API to generate an in-dialog UAC – this could be perfect…

But alas real life came back to rear its ugly head, and this adventure will have to continue another day…

Update: Thanks to a kindly provided PCAP I now know what I was doing wrong, and so we’ll soon have a follow up to this post named “Successes in cobbling together a USSD Gateway” just as soon as I have a weekend free.

SMS-over-IP Message Efficiency – K

Recently I read a post from someone talking about efficiency of USSD over IMS, and how crazy it was that such a small amount of data used so much overhead to get transferred across the network.

Having built an SMSc a while ago, my mind immediately jumped to SMS over IMS as being a great example of having so much overhead.

If we’re to consider sending the response “K” to a text message, how much overhead is there?

SMS PDU containing the message “K”

I’m using a common Qualcomm based smartphone, and here’s the numbers I’ve got from Wireshark when I send the message:

Transport Ethernet Header – 14 Bytes
Transport IP Header – 20 Bytes
Transport UDP Header – 8 Bytes
Transport GTP Header – 12 Bytes
User IP Header – 20 Bytes
IPsec ESP Header (For Um interface protection) – 22 Bytes
Encapsulated UDP Header – 8 Bytes
SIP Headers – 707 Bytes
SMS Header – 16 Bytes
SMS Message Body “K” – 1 Byte

Overall SIP, ESP, GTP and Transport PCAP for SIP MESSAGE

That seems pretty bad in terms of efficiency, but let’s look at how that actually works out:

This means our actual message body makes up just 1 byte of 828 bytes, or 0.12% of the size of the overall payload.

Even combined with the SMS header (which contains all the addressing information needed to route an SMS) it’s still just on 2% of the overall message.

So USSD efficiency isn’t great, but it’s not alone!

Diameter Routing Agents – Part 5 – AVP Transformations with FreeDiameter and Python in rt_pyform

In our last post we talked about why we’d want to perform Diameter AVP translations / rewriting on our Diameter Routing Agent.

Now let’s look at how we can actually achieve this using rt_pyform extension for FreeDiameter and some simple Python code.

Before we build we’ll need to make sure we have the python3-devel package (I’m using python3-devel-3.10) installed.

Then we’ll build FreeDiameter with the rt_pyform, this branch contains the rt_pyform extension in it already, or you can clone the extension only from this repo.

Now once FreeDiameter is installed we can load the extension in our freeDiameter.conf file:

LoadExtension = "rt_pyform.fdx" : "<Your config filename>.conf";

Next we’ll need to define our rt_pyform config, this is a super simple 3 line config file that specifies the path of what we’re doing:

DirectoryPath = "."        # Directory to search
ModuleName = "script"      # Name of python file. Note there is no .py extension
FunctionName = "transform" # Python function to call

The DirectoryPath directive specifies where we should search for the Python code, and ModuleName is the name of the Python script, lastly we have FunctionName which is the name of the Python function that does the rewriting.

Now let’s write our Python function for the transformation.

The Python function much have the correct number of parameters, must return a string, and must use the name specified in the config.

The following is an example of a function that prints out all the values it receives:

def transform(appId, flags, cmdCode, HBH_ID, E2E_ID, AVP_Code, vendorID, value):
    print('[PYTHON]')
    print(f'|-> appId: {appId}')
    print(f'|-> flags: {hex(flags)}')
    print(f'|-> cmdCode: {cmdCode}')
    print(f'|-> HBH_ID: {hex(HBH_ID)}')
    print(f'|-> E2E_ID: {hex(E2E_ID)}')
    print(f'|-> AVP_Code: {AVP_Code}')
    print(f'|-> vendorID: {vendorID}')
    print(f'|-> value: {value}')
    
    return value

Note the order of the arguments and that return is of the same type as the AVP value (string).

We can expand upon this and add conditionals, let’s take a look at some more complex examples:

def transform(appId, flags, cmdCode, HBH_ID, E2E_ID, AVP_Code, vendorID, value):
    print('[PYTHON]')
    print(f'|-> appId: {appId}')
    print(f'|-> flags: {hex(flags)}')
    print(f'|-> cmdCode: {cmdCode}')
    print(f'|-> HBH_ID: {hex(HBH_ID)}')
    print(f'|-> E2E_ID: {hex(E2E_ID)}')
    print(f'|-> AVP_Code: {AVP_Code}')
    print(f'|-> vendorID: {vendorID}')
    print(f'|-> value: {value}')
    #IMSI Translation - if App ID = 16777251 and the AVP being evaluated is the Username
    if (int(appId) == 16777251) and int(AVP_Code) == 1:
        print("This is IMSI '" + str(value) + "' - Evaluating transformation")
        print("Original value: " + str(value))
        value = str(value[::-1]).zfill(15)

The above look at if the App ID is S6a, and the AVP being checked is AVP Code 1 (Username / IMSI ) and if so, reverses the username, so IMSI 1234567 becomes 7654321, the zfill is just to pad with leading 0s if required.

Now let’s do another one for a Realm Rewrite:

def transform(appId, flags, cmdCode, HBH_ID, E2E_ID, AVP_Code, vendorID, value):

    #Print Debug Info
    print('[PYTHON]')
    print(f'|-> appId: {appId}')
    print(f'|-> flags: {hex(flags)}')
    print(f'|-> cmdCode: {cmdCode}')
    print(f'|-> HBH_ID: {hex(HBH_ID)}')
    print(f'|-> E2E_ID: {hex(E2E_ID)}')
    print(f'|-> AVP_Code: {AVP_Code}')
    print(f'|-> vendorID: {vendorID}')
    print(f'|-> value: {value}')
    #Realm Translation
    if int(AVP_Code) == 283:
        print("This is Destination Realm '" + str(value) + "' - Evaluating transformation")
    if value == "epc.mnc001.mcc001.3gppnetwork.org":
        new_realm = "epc.mnc999.mcc999.3gppnetwork.org"
        print("translating from " + str(value) + " to " + str(new_realm))
        value = new_realm
    else:
        #If the Realm doesn't match the above conditions, then don't change anything
        print("No modification made to Realm as conditions not met")
    print("Updated Value: " + str(value))

In the above block if the Realm is set to epc.mnc001.mcc001.3gppnetwork.org it is rewritten to epc.mnc999.mcc999.3gppnetwork.org, hopefully you can get a handle on the sorts of transformations we can do with this – We can translate any string type AVPs, which allows for hostname, realm, IMSI, Sh-User-Data, Location-Info, etc, etc, to be rewritten.

Diameter Routing Agents – Part 5 – AVP Transformations

Having a central pair of Diameter routing agents allows us to drastically simplify our network, but what if we want to perform some translations on AVPs?

For starters, what is an AVP transformation? Well it’s simply rewriting the value of an AVP as the Diameter Request/Response passes through the DRA. A request may come into the DRA with IMSI xxxxxx and leave with IMSI yyyyyy if a translation is applied.

So why would we want to do this?

Well, what if we purchased another operator who used Realm X, and we use Realm Y, and we want to link the two networks, then we’d need to rewrite Realm Y to Realm X, and Realm X to Realm Y when they communicate, AVP transformations allow for this.

If we’re an MVNO with hosted IMSIs from an MNO, but want to keep just the one IMSI in our HSS/OCS, we can translate from the MNO hosted IMSI to our internal IMSI, using AVP transformations.

If our OCS supports only one rating group, and we want to rewrite all rating groups to that one value, AVP transformations cover this too.

There are lots of uses for this, and if you’ve worked with a bit of signaling before you’ll know that quite often these sorts of use-cases come up.

So how do we do this with freeDiameter?

To handle this I developed a module for passing each AVP to a Python function, which can then apply any transformation to a text based value, using every tool available to you in Python.

In the next post I’ll introduce rt_pyform and how we can use it with Python to translate Diameter AVPs.

Diameter Routing Agents – Part 4 – Advanced FreeDiameter DRA Routing

Way back in part 2 we discussed the basic routing logic a DRA handles, but what if we want to do something a bit outside of the box in terms of how we route?

For me, one of the most useful use cases for a DRA is to route traffic based on IMSI / Username.
This means I can route all the traffic for MVNO X to MVNO X’s HSS, or for staging / test subs to the test HSS enviroment.

FreeDiameter has a bunch of built in logic that handles routing based on a weight, but we can override this, using the rt_default module.

In our last post we had this module commented out, but let’s uncomment it and start playing with it:

#Basic Diameter config for this box
Identity = "dra.mnc001.mcc001.3gppnetwork.org";
Realm = "mnc001.mcc001.3gppnetwork.org";
Port = 3868;

LoadExtension = "dbg_msg_dumps.fdx" : "0x8888";
LoadExtension = "rt_redirect.fdx":"0x0080";
LoadExtension = "rt_default.fdx":"rt_default.conf";

TLS_Cred = "/etc/freeDiameter/cert.pem", "/etc/freeDiameter/privkey.pem";
TLS_CA = "/etc/freeDiameter/cert.pem";
TLS_DH_File = "/etc/freeDiameter/dh.pem";

ConnectPeer = "mme01.mnc001.mcc001.3gppnetwork.org" { ConnectTo = "10.98.0.10"; No_TLS; };
ConnectPeer = "hss01" { ConnectTo = "10.0.1.252"; No_TLS; Port = 3868; Realm = "mnc001.mcc001.3gppnetwork.org";};
ConnectPeer = "hss02" { ConnectTo = "10.0.1.253"; No_TLS; Port = 3868; Realm = "mnc001.mcc001.3gppnetwork.org";};
ConnectPeer = "hss-mvno-x" { ConnectTo = "10.98.0.22"; No_TLS; Port = 3868; Realm = "mnc001.mcc001.3gppnetwork.org";};
ConnectPeer = "hss-lab" { ConnectTo = "10.0.2.2"; No_TLS; Port = 3868; Realm = "mnc001.mcc001.3gppnetwork.org";};

In the above code we’ve uncommented rt_default and rt_redirect.

You’ll notice that rt_default references a config file, so we’ll create a new file in our /etc/freeDiameter directory called rt_default.conf, and this is where the magic will happen.

A few points before we get started:

  • This overrides the default routing priorities, but in order for a peer to be selected, it has to be in an Open (active) state
  • The peer still has to have advertised support for the requested application in the CER/CEA dialog
  • The peers will still need to have all been defined in the freeDiameter.conf file in order to be selected

So with that in mind, and the 5 peers we have defined in our config above (assuming all are connected), let’s look at some rules we can setup using rt_default.

Intro to rt_default Rules

The rt_default.conf file contains a list of rules, each rule has a criteria that if matched, will result in the specified action being taken. The actions all revolve around how to route the traffic.

So what can these criteria match on?
Here’s the options:

Item to MatchCode
Any*
Origin-Hostoh=”STR/REG”
Origin-Realmor=”STR/REG”
Destination-Hostdh=”STR/REG”
Destination-Realmdr=”STR/REG”
User-Nameun=”STR/REG”
Session-Idsi=”STR/REG”
rt_default Matching Criteria

We can either match based on a string or a regex, for example, if we want to match anything where the Destination-Realm is “mnc001.mcc001.3gppnetwork.org” we’d use something like:

#Low score to HSS02
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss02" += -70 ;

Now you’ll notice there is some stuff after this, let’s look at that.

We’re matching anything where the destination-host is set to hss02 (that’s the bit before the colon), but what’s the bit after that?

Well if we imagine that all our Diameter peers are up, when a message comes in with Destination-Realm “mnc001.mcc001.3gppnetwork.org”, looking for an HSS, then in our example setup, we have 4 HHS instances to choose from (assuming they’re all online).

In default Diameter routing, all of these peers are in the same realm, and as they’re all HSS instances, they all support the same applications – Our request could go to any of them.

But what we set in the above example is simply the following:

If the Destination-Realm is set to mnc001.mcc001.3gppnetwork.org, then set the priority for routing to hss02 to the lowest possible value.

So that leaves the 3 other Diameter peers with a higher score than HSS02, so HSS02 won’t be used.

Let’s steer this a little more,

Let’s specify that we want to use HSS01 to handle all the requests (if it’s available), we can do that by adding a rule like this:

#Low score to HSS02
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss02" += -70 ;
#High score to HSS01
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss01" += 100 ;

But what if we want to route to hss-lab if the IMSI matches a specific value, well we can do that too.

#Low score to HSS02
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss02" += -70 ;
#High score to HSS01
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss01" += 100 ;
#Route traffic for IMSI to Lab HSS
un="001019999999999999" : dh="hss-lab" += 200 ;

Now that we’ve set an entry with a higher score than hss01 that will be matched if the username (IMSI) equals 001019999999999999, the traffic will get routed to hss-lab.

But that’s a whole IMSI, what if we want to match only part of a field?

Well, we can use regex in the Criteria as well, so let’s look at using some Regex, let’s say for example all our MVNO SIMs start with 001012xxxxxxx, let’s setup a rule to match that, and route to the MVNO HSS with a higher priority than our normal HSS:

#Low score to HSS02
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss02" += -70 ;
#High score to HSS01
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss01" += 100 ;
#Route traffic for IMSI to Lab HSS
un="001019999999999999" : dh="hss-lab" += 200 ;
#Route traffic where IMSI starts with 001012 to MVNO HSS
un=["^001012.*"] : dh="hss-mvno-x" += 200 ;

Let’s imagine that down the line we introduce HSS03 and HSS04, and we only want to use HSS01 if HSS03 and HSS04 are unavailable, and only to use HSS02 no other HSSes are available, and we want to split the traffic 50/50 across HSS03 and HSS04.

Firstly we’d need to add HSS03 and HSS04 to our FreeDiameter.conf file:

...
ConnectPeer = "hss02" { ConnectTo = "10.0.1.253"; No_TLS; Port = 3868; Realm = "mnc001.mcc001.3gppnetwork.org";};
ConnectPeer = "hss03" { ConnectTo = "10.0.3.3"; No_TLS; Port = 3868; Realm = "mnc001.mcc001.3gppnetwork.org";};
ConnectPeer = "hss04" { ConnectTo = "10.0.4.4"; No_TLS; Port = 3868; Realm = "mnc001.mcc001.3gppnetwork.org";};
...

Then in our rt_default.conf we’d need to tweak our scores again:

#Low score to HSS02
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss02" += 10 ;
#Medium score to HSS01
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss01" += 20 ;
#Route traffic for IMSI to Lab HSS
un="001019999999999999" : dh="hss-lab" += 200 ;
#Route traffic where IMSI starts with 001012 to MVNO HSS
un=["^001012.*"] : dh="hss-mvno-x" += 200 ;
#High Score for HSS03 and HSS04
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss02" += 100 ;
dr="mnc001.mcc001.3gppnetwork.org" : dh="hss04" += 100 ;

One quick tip to keep your logic a bit simpler, is that we can set a variety of different values based on keywords (listed below) rather than on a weight/score:

BehaviourNameScore
Do not deliver to peer (set lowest priority)NO_DELIVERY-70
The peer is a default route for all messagesDEFAULT5
The peer is a default route for this realmDEFAULT_REALM10
REALM15
Route to the specified Host with highest priorityFINALDEST100
Rather than manually specifying the store you can use keywords like above to set the value

In our next post we’ll look at using FreeDiameter based DRA in roaming scenarios where we route messages across Diameter Realms.

Diameter Routing Agents (Why you need them, and how to build them) – Part 2 – Routing

What I typically refer to as Diameter interfaces / reference points, such as S6a, Sh, Sx, Sy, Gx, Gy, Zh, etc, etc, are also known as Applications.

Diameter Application Support

If you look inside the Capabilities Exchange Request / Answer dialog, what you’ll see is each side advertising the Applications (interfaces) that they support, each one being identified by an Application ID.

CER showing support for the 3GPP Zh Application-ID (Interface)

If two peers share a common Application-Id, then they can communicate using that Application / Interface.

For example, the above screenshot shows a peer with support for the Zh Interface (Spoiler alert, XCAP Gateway / BSF coming soon!). If two Diameter peers both have support for the Zh interface, then they can use that to send requests / responses to each other.

This is the basis of Diameter Routing.

Diameter Routing Tables

Like any router, our DRA needs to have logic to select which peer to route each message to.

For each Diameter connection to our DRA, it will build up a Diameter Routing table, with information on each peer, including the realm and applications it advertises support for.

Then, based on the logic defined in the DRA to select which Diameter peer to route each request to.

In its simplest form, Diameter routing is based on a few things:

  1. Look at the DestinationRealm, and see if we have any peers at that realm
  2. If we do then look at the DestinationHost, if that’s set, and the host is connected, and if it supports the specified Application-Id, then route it to that host
  3. If no DestinationHost is specified, look at the peers we have available and find the one that supports the specified Application-Id, then route it to that host
Simplified Diameter Routing Table used by DRAs

With this in mind, we can go back to looking at how our DRA may route a request from a connected MME towards an HSS.

Let’s look at some examples of this at play.

The request from MME02 is for DestinationRealm mnc001.mcc001.3gppnetwork.org, which our DRA knows it has 4 connected peers in (3 if we exclude the source of the request, as we don’t want to route it back to itself of course).

So we have 3 contenders still for who could get the request, but wait! We have a DestinationHost specified, so the DRA confirms the host is available, and that it supports the requested ApplicationId and routes it to HSS02.

So just because we are going through a DRA does not mean we can’t specific which destination host we need, just like we would if we had a direct link between each Diameter peer.

Conversely, if we sent another S6a request from MME01 but with no DestinationHost set, let’s see how that would look.

Again, the request is from MME02 is for DestinationRealm mnc001.mcc001.3gppnetwork.org, which our DRA knows it has 3 other peers it could route this to. But only two of those peers support the S6a Application, so the request would be split between the two peers evenly.

Clever Routing with DRAs

So with our DRA in place we can simplify the network, we don’t need to build peer links between every Diameter device to every other, but let’s look at some other ways DRAs can help us.

Load Control

We may want to always send requests to HSS01 and only use HSS02 if HSS01 is not available, we can do this with a DRA.

Or we may want to split load 75% on one HSS and 25% on the other.

Both are great use cases for a DRA.

Routing based on Username

We may want to route requests in the DRA based on other factors, such as the IMSI.

Our IMSIs may start with 001010001xxx, but if we introduced an MVNO with IMSIs starting with 001010002xxx, we’d need to know to route all traffic where the IMSI belongs to the home network to the home network HSS, and all the MVNO IMSI traffic to the MVNO’s HSS, and DRAs handle this.

Inter-Realm Routing

One of the main use cases you’ll see for DRAs is in Roaming scenarios.

For example, if we have a roaming agreement with a subscriber who’s IMSIs start with 90170, we can route all the traffic for their subs towards their HSS.

But wait, their Realm will be mnc901.mcc070.3gppnetwork.org, so in that scenario we’ll need to add a rule to route the request to a different realm.

DRAs handle this also.

In our next post we’ll start actually setting up a DRA with a default route table, and then look at some more advanced options for Diameter routing like we’ve just discussed.

One slight caveat, is that mutual support does not always mean what you may expect.
For example an MME and an HSS both support S6a, which is identified by Auth-Application-Id 16777251 (Vendor ID 10415), but one is a client and one is a server.
Keep this in mind!

Kamailio I-CSCF – SRV Lookup Behaviour

Recently I had a strange issue I thought I’d share.

Using Kamailio as an Interrogating-CSCF, Kamailio was getting the S-CSCF details from the User-Authorization-Answer’s “Server-Name” (602) AVP.

The value was set to:

sip:scscf.mnc001.mcc001.3gppnetwork.org:5060

But the I-CSCF was only looking up A-Records for scscf.mnc001.mcc001.3gppnetwork.org, not using DNS-SRV.

The problem? The Server-Name I had configured as a full SIP URI in PyHSS including the port, meant that Kamailio only looks up the A-Record, and did not do a DNS-SRV lookup for the domain.

Dropping the port number saw all those delicious SRV records being queried.

Something to keep in mind if you use S-CSCF pooling with a Kamailio based I-CSCF, if you want to use SRV records for load balancing / traffic sharing, don’t include the port, and if instead you want it to go to the specified host found by an A-record, include the port.

SMS with Alphanumeric Source

Sending SMS with an alphanumeric String as the Source

If you’ve ever received an SMS from your operator, and the sender was the Operator name for example, you may be left wondering how it’s done.

In IMS you’d think this could be quite simple – You’d set the From header to be the name rather than the MSISDN, but for most SMSoIP deployments, the From header is ignored and instead the c header inside the SMS body is used.

So how do we get it to show text?

Well the TP-Originating address has the “Type of Number” (ToN) field which is typically set to International/National, but value 5 allows for the Digits to instead be alphanumeric characters.

GSM 7 bit encoding on the text in the TP-Originating Address digits and presto, you can send SMS to subscribers where the message shows as From an alphanumeric source.

On Android SMSs received from alphanumeric sources cannot be responded to (“no more “DO NOT REPLY TO THIS MESSAGE” at the end of each text), but on iOS devices you can respond, but if I send an SMS from “Nick” the reply from the subscriber using the iPhone will be sent to MSISDN 6425 (Nick on the telephone keypad).

FreeSWITCH – Incompatible Destination

A recent little issue I ran into the other day, that I figured may be of use to someone in the future.

When making a call to FreeSWITCH I would get an “INCOMPATIBLE DESTINATION” response to the SIP INVITE.

Here’s what I saw in the log:

FreeSWITCH showing an “INCOMPATIBLE DESTINATION” error
2022-02-19 13:04:04.027963 99.47% [DEBUG] switch_core_media.c:5650 Audio Codec Compare [GSM:3:8000:20:13200:1]/[opus:116:48000:20:0:1]
2022-02-19 13:04:04.027963 99.47% [DEBUG] switch_core_media.c:5650 Audio Codec Compare [GSM:3:8000:20:13200:1]/[G722:9:8000:20:64000:1]
2022-02-19 13:04:04.027963 99.47% [DEBUG] switch_core_media.c:5650 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1]
2022-02-19 13:04:04.027963 99.47% [DEBUG] switch_core_media.c:5650 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1]
2022-02-19 13:04:04.027963 99.47% [DEBUG] switch_core_media.c:5944 No 2833 in SDP. Liberal DTMF mode adding 101 as telephone-event.
2022-02-19 13:04:04.027963 99.47% [DEBUG] switch_core_media.c:5973 sofia/internal/[email protected]:5060 Set 2833 dtmf send payload to 101 recv payload to 101
2022-02-19 13:04:04.027963 99.47% [NOTICE] switch_channel.c:3993 Hangup sofia/internal/[email protected]:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

The hint to the cause of the error is above it – Codec comparison. If we look at the Audio Codec Compare lines, we can see the GSM codec we are trying to use, does not match the codecs configured in FreeSWITCH, hence getting the INCOMPATIBLE_DESTINATION error – None of the codecs offered match the codecs supported in FreeSWITCH.

So where do we go to fix this?

Well the SIP profile itself defines the codecs that are supported on this SIP profile,

FreeSWITCH SIP Profile (Sofia) codec settings

If you’re using a mostly default config, you’ll see this is set to a global variable, called $${global_codec_prefs}, so let’s take a look at vars.xml where this is defined:

FreeSWITCH default codec selection global variable

And there’s our problem, we need to add the GSM codec into that list to allow the calls,

So we change it to add the codecs we want to support, and reload the changes,

The Codec preferences I need for this IMS Application Server

Now when we want to make a call, success!

Successful call
IMS DNS Failing

Kamailio, IMS & DNS Headches

I’m sure I’ve ranted about the importance of DNS in IMS networks in the past on here already.

Recently I was rebuilding a P-CSCF and kept getting an error saying that the DNS was failing to resolve:

 4(5993) CRITICAL: <core> [core/dns_cache.c:3136]: dns_srv_sip_resolve(): unknown proto 0
 4(5993) ERROR: tm [ut.h:284]: uri2dst2(): failed to resolve "ims.mnc001.mcc001.3gppnetwork.org" :bug - critical error (-13)
 4(5993) ERROR: tm [t_fwd.c:1759]: t_forward_nonack(): failure to add branches
 4(5993) ERROR: sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: Unresolvable destination (478/SL)

This was a rebuild, another P-CSCF was running fine and handling traffic with the same DNS server set.

I checked the netplan config and confirmed the DNS server was set correctly.

If I did an nslookup on the address that was failing to resolve – pointing it at the correct DNS server, the A & SRV records came back OK, and everything was fine.

Stranger still, after clearing the DNS Cache, and running a packet capture, I couldn’t see any DNS queries at all….

The problem? Kamailio uses resolv.conf by default on Ubuntu Server, and that was pointing to localhost.

After updating resolv.conf to point to the DNS server handling the IMS domains, I was good to go again.

A super valuable resource for all things DNS & Kamailio is this doc.

FreeSWITCH, Kamailio & IMS Extensions

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.

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!

The Surprisingly Complicated World of SMS: Apple iPhone MT SMS

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.

Lifecycle of a Dedicated Bearer – From Flow-Description AVP to Traffic Flow Templates

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.

So within the Gx Re-Auth Request is the Charging-Rule Definition, made up of Flow-Description AVP which I’ve written about here, that is used to identify and profile traffic flows and QoS parameters to apply to matching traffic.

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
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!

Credit Control Request / Answer call flow in IMS Charging

Basics of EPC/LTE Online Charging (OCS)

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.

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.