Tag Archives: IMS

Tales from the Trenches – SMS Data Coding Scheme 0

The Data Coding Scheme (DCS or TP-DCS) header in an SMS body indicates what encoding is used in that message.

It means if we’re using UCS-2 (UTF16) special characters like Emojis etc, in in our message, the phone knows to decode the data in the message body using UTF, because the Data Coding Scheme (DCS) header indicates the contents are encoded in UTF.

Likewise, if we’re not using any fancy characters in our message and the message is encoded as plain old GSM7, we set set the DCS to 0 to indicate this is using GSM7.

From my experience, I’d always assumed that DCS0 (Default) == GSM7, but today I learned, that’s not always the case. Some SMSc entities treat DCS0 as Latin.

Let me explain why this is stupid and why I wasted a lot of time on this.

We can indicate that a message is encoded as Latin by setting the DCS to 0x03:

We cannot indicate that the message is encoded as GSM7 through anything other than the default alphabet (DCS 0).

Latin has it’s own encoding flag, if I wanted the message treated as Latin, I’d indicate the message encoding is Latin in the DCS bit!

I spent a bunch of time trying to work out why a customer was having issues getting messages to subscribers on another operator, and it turned out the other operator treats messages we send to them on SMPP with DCS0 as Latin encoding, and then cracks the sads when trying to deliver it.

The above diff shows the message we send (Right), and the message they dry to deliver (left).

Well, lesson learned…

Creating a Fixed Line IMS subscriber in PyHSS

I generally do this with Python or via the Swagger UI for the Web UI, but here’s how we can create a fixed-line IMS subscriber in PyHSS, so we can register it with a softphone, without using EAP-AKA.


Firstly we create the AuC object for this password combo.

curl -X 'PUT' \
'http://10.97.0.36:8080/auc/' \
-H 'accept: application/json' \
-H 'Content-Type: application/json' \
-d '{
"ki": "yoursippassword123",
"opc": "",
"amf": "8000",
"sqn": "1"
}'

Get back the AuC ID from the JSON body, we’ll use this to provision the Sub:

 "auc_id": 110,

Next we create the subscriber, the speeds will be 0 as there is no data on the service, but we will add an default APN so the validation passes:

curl -X 'PUT' \
'http://10.97.0.36:8080/subscriber/' \
-H 'accept: application/json' \
-H 'Content-Type: application/json' \
-d '{
"default_apn": 1,
"roaming_rule_list": null,
"apn_list": "0",
"subscribed_rau_tau_timer": 600,
"msisdn": "123451000001",
"ue_ambr_dl": 0,
"ue_ambr_ul": 0,
"imsi": "001001000090001",
"nam": 2,
"enabled": true,
"roaming_enabled": null,
"auc_id": 110
}'

Alright, that’s the basics done, now we’ll create the IMS subscriber.

Provision it:

curl -X 'PUT' \
'http://10.97.0.36:8080/ims_subscriber/' \
-H 'accept: application/json' \
-H 'Content-Type: application/json' \
-d '{
"pcscf_realm": null,
"scscf_realm": null,
"pcscf_active_session": null,
"scscf_peer": null,
"msisdn": "123451000001",
"pcscf_timestamp": null,
"sh_template_path": "default_sh_user_data.xml",
"msisdn_list": "123451000001",
"pcscf_peer": null,
"last_modified": "2024-04-25T00:33:37Z",
"imsi": "001001000090001",
"xcap_profile": null,
"ifc_path": "default_ifc.xml",
"sh_profile": "\n<!-- This container for the XCAP Data for the Subscriber -->\n<RepositoryData>\n <ServiceIndication>ApplicationServer</ServiceIndication>\n <SequenceNumber>0</SequenceNumber>\n <ServiceData>\n <!-- This is the actual XCAP Data for the Subscriber -->\n \n <!-- XCAP Default Template (no XCAP Data stored in Database) -->\n <simservs xmlns=\"http://uri.etsi.org/ngn/params/xml/simservs/xcap\" xmlns:cp=\"urn:ietf:params:xml:ns:common-policy\">\n <originating-identity-presentation active=\"true\" />\n <originating-identity-presentation-restriction active=\"true\">\n <default-behaviour>presentation-not-restricted</default-behaviour>\n </originating-identity-presentation-restriction>\n <communication-diversion active=\"true\">\n <!-- No Answer Time -->\n <NoReplyTimer>20</NoReplyTimer>\n <cp:ruleset>\n <!-- Call Forward All Rule -->\n <cp:rule id=\"rule0\">\n <cp:conditions>\n <communication-diverted />\n </cp:conditions>\n <cp:actions>\n <forward-to>\n <target>sip:[email protected]</target>\n </forward-to>\n </cp:actions>\n </cp:rule>\n <!-- Call Forward Not Registered Rule -->\n <cp:rule id=\"rule1\">\n <cp:conditions>\n <not-registered />\n </cp:conditions>\n <cp:actions>\n <forward-to>\n <target>sip:[email protected]</target>\n </forward-to>\n </cp:actions>\n </cp:rule>\n <!-- Call Forward No Answer Rule -->\n <cp:rule id=\"rule2\">\n <cp:conditions>\n <no-answer />\n </cp:conditions>\n <cp:actions>\n <forward-to>\n <target>sip:[email protected]</target>\n </forward-to>\n </cp:actions>\n </cp:rule>\n <!-- Call Forward Busy Rule -->\n <cp:rule id=\"rule3\">\n <cp:conditions>\n <busy />\n </cp:conditions>\n <cp:actions>\n <forward-to>\n <target>sip:[email protected]</target>\n </forward-to>\n </cp:actions>\n </cp:rule>\n <!-- Call Forward Unreachable Rule -->\n <cp:rule id=\"rule4\">\n <cp:conditions>\n <not-reachable />\n </cp:conditions>\n <cp:actions>\n <forward-to>\n <target>sip:[email protected]</target>\n </forward-to>\n </cp:actions>\n </cp:rule>\n </cp:ruleset>\n </communication-diversion>\n \n <incoming-communication-barring active=\"true\">\n <cp:ruleset>\n <cp:rule id=\"rule0\">\n <cp:conditions />\n <cp:actions>\n <allow>true</allow>\n </cp:actions>\n </cp:rule>\n </cp:ruleset>\n </incoming-communication-barring>\n\n <outgoing-communication-barring active=\"false\">\n </outgoing-communication-barring>\n </simservs>\n\n </ServiceData>\n \n</RepositoryData>\n",
"pcscf": null,
"scscf": null,
"scscf_timestamp": null
}'

And with that we’re done,

We can now register 001001000090001 at our IMS, with password yoursippassword123 which has the MSISDN / phone number 123451000001.

Easy!

Getting started with PyHSS

PyHSS is our open source Home Subscriber Server, it’s written in Python, has a variety of different backends, and is highly perforate (We benchmark to 10K transactions per second) and infinitely scaleable.

In this post I’ll cover the basics of setting up PyHSS in your enviroment and getting some Diameter peers connected.

For starters, we’ll need a database (We’ll use MySQL for this demo) and an account on that database for a MySQL user.

So let’s get that rolling (I’m using Ubuntu 24.04):

sudo apt update
sudo apt install mysql-server

Next we’ll create the MySQL user for PyHSS to use:

CREATE USER 'pyhss_user'@'%' IDENTIFIED BY 'pyhss_password';
GRANT ALL PRIVILEGES ON *.* TO 'pyhss_user'@'%' WITH GRANT OPTION;
FLUSH PRIVILEGES;

We’ll also need Redis as well (PyHSS uses Redis for inter-service communications and for caching), so go ahead an install that for your distro:

sudo apt install redis-server

So that’s our prerequisites sorted, let’s clone the PyHSS repo:

git clone https://github.com/nickvsnetworking/pyhss /etc/pyhss

And install the requirements with pip from the PyHSS repo:

pip3 install -r requirements.txt

Next we’ll need to configure PyHSS, for that we update the config file (config.yaml) with the settings we want to use.

We’ll start by setting the bind_ip to a list of IPs you want to listen on, and your transport – We can use either TCP or SCTP.

For Diameter, we will set OriginHost and OriginRealm to match the Diameter hostname you want to use for this peer, and the Realm of your Diameter network.

Lastly we’ll need to set the database parameters, updating the database: section to populate your credentials, setting your username and password and the database to match your SQL installation we setup at the start.

With that done, we can start PyHSS, which we do using systemctl.

Because there’s multiple microservices that make up PyHSS, there’s multiple systemctl files use to run PyHSS as a service, they’re all in the /systemd folder.

We’ll copy them all to our systemd folder.

cp /etc/pyhss/systemd/ /lib/systemd/system/
systemctl daemon-reload
systemctl start pyhss

And with that we’ve got PyHSS running and ready for a Diameter peer to connect.

Now you should be able to bring our Diameter peers up.

If you’re using something like Kamailio, with C-Diameter Peer, you can read about the config for that here, or FreeDiameter you can read about here.

In the next post, we’ll cover subscriber provisioning via the API.

The power of the PyHSS EIR

The Equipment Identity Register (EIR) is a pretty handy function in 3GPP networks.

Via the Diameter based S13 interface, the MME, is able to query the EIR to ask if a given IMEI & IMSI combination should be allowed to attach.

This allows stolen / grey market / unauthorized devices (IMEIs) to be rejected from the network, the EIR can have a list of “bad” IMEIs that if seen will reject the request.

It also allows us to lock a SIM (IMSI) to a given device (IMEI) or type of device – We can use this for say a Fixed Wireless service, to lock the SIMs (IMSIs) to a range of modems (IMEI Prefixes).

Lastly it gives us insight and analytics into the devices used on the network, by mapping the IMEI to a device, we can say that IMEI 1234567890 is an Apple iPhone 12 Pro Max, or a Nokia Fastmile 5G-24W-A.

PyHSS supports all these capabilities, so let’s have a look at how we’d manage / access them.

Setting up EIR Rules

These rules are set via the RESTful API in PyHSS.

The Equipment Identity Register built into PyHSS supports matching in one of two modes, set by regex_mode.

In Exact Mode (regex_mode: 0) matches are based on an exact matching IMEI, and matching the IMSI if set (If IMSI is set to nothing (”), then only the IMEI is evaluated).

Exact Mode is suited for IMEI/IMSI locking, to ensure a SIM is locked to a particular device, or to blacklist stolen devices.

Regex Mode (regex_mode: 1) matches based on Regex, this is suited for whitelisting IMEI prefixes for say, specific validated vendors.

The match_response_code maps to the Equipment-Status AVP output, so specified values are:

  • 0 : ‘Whitelist’
  • 1: ‘Blacklist’
  • 2: ‘Greylist’

Some end to end examples of this provisioned into the API:

IMSI / IMEI Binding

{
      'imei': '1234', 
      'imsi': '567',
      'regex_mode': 0, 
      'match_response_code': 0
}

If IMSI is equal to 567 and is in use in IMEI 1234, then the response code returned is 0 (Whitelist).

IMEI Matching (Blacklist lost / stolen devices)

{
      'imei': '99881232',
      'imsi': '', 
      'regex_mode': 0, 
      'match_response_code': 1
}

If the IMEI is equal to 99881232 used with any IMSI, then the response code returned is 1 (Blacklist). This would be used for devices reported stolen.

IMEI Prefix Match (Blacklist / Whitelist all devices of type)

{
      'imei': '^666.*',
      'imsi': '', 
      'regex_mode': 1, 
      'match_response_code': 1
}

If the IMEI starts with 666, then the response code returned is 1 (Blacklist).

IMEI & IMSI Regex Match

{
      'imei': '^777.*',
      'imsi': '^1234123412341234$', 
      'regex_mode': 1, 
      'match_response_code': 2
}

If the IMEI starts with 777 and the IMSI is 1234123412341234 then return 2 (Greylist).

No Match Behaviour

If there is no match from the backend, then the config parameter no_match_response dictates the response code returned (Blacklist/Whitelist/Greylist).

Mapping Type Allocation Codes (TACs) to IMEIs

There are several data feeds of the Type Allocation Codes (TACs) which map a given IMEI prefix to a model number.

TAC database extract

Unfortunately, this data is not freely available, so we can’t bundle it with PyHSS, but if you have the IMEI Database, you can load it into PyHSS using Redis, to allow us to report on this data.

In your config.yaml you’ll just need to set the tac_database parameter, which will read the data on startup.

PyHSS YAML Config extract

Triggering on SIM Swap

If we keep track of the current IMSI/IMEI combination used for each SIM/Device, we can get notified every time it changes.

You might want to use this to trigger OTA provisioning or clear old data in your IMS.

For that we can use the sim_swap_notify_webhook in the config to send a HTTP POST to a given endpoint to inform it that a SIM is now in a different device.

We also have to have imsi_imei_logging set to true in the Config in order to log the history.

Reporting on IMEIs

We can also log/capture historical data about IMSI/IMEI combinations.

We use this from a customer support perspective to be able to see if a customer has recently changed phones, so if they call support, our staff can ask the customer about it to help troubleshoot.

“I can see you were connected previously on a Samsung Galaxy S22, but now you’re using a Nokia 3310, did the issues happen before you moved phones?”

This is super handy.

We can get a general log of IMSI vs IMEI like this:

Feed of IMSI vs IMEI along with a timestamp and the response that was sent back

But what’s more useful is searching for a IMSI or an IMEI and then getting back a full list of devices / SIMs that have been used.

Searching for an IMSI I can see it’s only ever been used in this Samsung Galaxy

Lastly via Grafana we export all this data, which allows us to visualize this data and build dashboards showing the devices on the network.

Visualizing EIR Data in Grafana

PyHSS includes a Promethues exporter, when it comes to prom_eir_devices_total it lists each seen Type Allocation Code / UE in the network, along with the number we’ve seen of each.

Raw it looks like this:

But visualized in Grafana we can get a dashboard to give us a breakdown per vendor:

How do you know if they’re roaming? Charging challenges in IMS for Roamers

I got an email the other day asking a simple question:

How do I know if a subscriber is VoLTE roaming or not when they send an SMS to charge for it?

My immediate reaction was to look at the SIP headers, P-Access-Network-Info will tell you where the subscriber is located, end of.

Right?

Well not quite, this will tell the SMSc the location of the subscriber sending the SMS. If the PLMN in the P-Access-Network-Info != the home PLMN, the sub is roaming.

But does this information get passed to the OCS / OFCS?

The SMSc uses “Event based charging” to perform credit control, so let’s have a look at what AVPs are present in the Credit Control Request from the SMSc:

Hmm, the SMS-Information AVP (2000) contains a bunch of information about the SMS being sent, but I don’t see anything about the location of the sender in there.

Originator-Interface is just set to “SIP”, of course in a 2G/3G roaming scenario the Originator-SCCP-Address would be that of the Visited PLMN, but for us it is our SCCP address.

Maybe the standard allows for an additional optional AVP in the SMS-Information-AVP we’re missing? Let’s check TS 32.299:

Nope.

So how to deal with this?

While the standards aren’t totally clear on this, we added an IMS-Info AVP and inside that populated the Access-Network-Information directly from the SIP header, and then picked that off inside our OCS in order to apply the correct rules.

Kamailio Bytes: Stripping SIP Multipart Bodies

For some calls in (such as some IMS emergency calls) you’ll get MIME Multipart Media Encapsulation as the SIP body, as the content-type set to:

Content-Type: multipart/mixed;boundary=968f194ab800ab27

If you’re used to dealing with SIP, you’d expect to see:

Content-Type: application/sdp

This Content-Type multipart/mixed;boundary is totally valid in SIP, in fact RFC 5261 (Message Body Handling in the Session Initiation Protocol (SIP)) details the use of MIME in SIP, and the Geolocation extension uses this, as we see below from a 911 call example.

But while this extension is standardised, and having your SIP Body containing multipart MIME is legal, not everything supports this, including the FreeSWITCH bridge module, which just appends a new SDP body into the Mime Multipart

Site note: I noticed FreeSWITCH Bridge function just appends the new SIP body in the multipart MIME, leaving the original, SDP:

Okay, so how do we replace the MIME Multipart SIP body with a standard SDP?

Well, with Kamalio’s SDP Ops Module, it’s fairly easy:

#If the body is multipart then strip it and replace with a single part
if (has_body("multipart/mixed")) {
	xlog("This has a multipart body");
	if (filter_body("application/sdp")) {
		remove_hf("Content-Type");
		append_hf("Content-Type: application/sdp\r\n");
	} else {
		xlog("Body part application/sdp not found\n");
		}
}

I’ve written about using SDPops to modify SDP before.

And with that we’ll take an SIP message like the one shown on the left, and when relayed, end up with the message on the right:

Simple fix, but saved me having to fix the fault in FreeSWITCH.

Android and Emergency Calling

In the last post we looked at emergency calling when roaming, and I mentioned that there are databases on the handsets for emergency numbers, to allow for example, calling 999 from a US phone, with a US SIM, roaming into the UK.

Android, being open source, allows us to see how this logic works, and it’s important for operators to understand this logic, as it’s what dictates the behavior in many scenarios.

It’s important to note that I’m not covering Apple here, this information is not publicly available to share for iOS devices, so I won’t be sharing anything on this – Apple has their own ecosystem to handle emergency calling, if you’re from an operator and reading this, I’d suggest getting in touch with your Apple account manager to discuss it, they’re always great to work with.

The Android Open Source Project has an “emergency number database”. This database has each of the emergency phone numbers and the corresponding service, for each country.

This file can be read at packages/services/Telephony/ecc/input/eccdata.txt on a phone with engineering mode.

Let’s take a look what’s in mainline Android for Australia:

You can check ECC for countries from the database on the AOSP repo.

This is one of the ways handsets know what codes represent emergency calling codes in different countries, alongside the values set in the SIM and provided by the visited network.

Tales from the Trenches: mode-set in AMR

This one was a bit of a head scratcher for me, but I’m always glad to learn something new.

The handset made a VoLTE call, and it’s SDP offer shows it can support AMR and AMR-WB:

        Media Attribute (a): rtpmap:116 AMR-WB/16000/1
        Media Attribute (a): fmtp:116 mode-set=0,1,2,3,4,5,6,7,8;mode-change-capability=2;max-red=220
        Media Attribute (a): rtpmap:118 AMR/8000/1
        Media Attribute (a): fmtp:118 mode-set=0,1,2,3,4,5,6,7;mode-change-capability=2;max-red=220
        Media Attribute (a): rtpmap:111 telephone-event/16000
        Media Attribute (a): fmtp:111 0-15

Okay, that’s pretty normal, I can see we have the mode-set parameter defined, which indicates what modes the handset supports for each codec.

In our problem scenario, the Media Gateway that the call was sent to responded with this SDP answer:

        Media Description, name and address (m): audio 24504 RTP/AVP 118 110
        Media Attribute (a): rtpmap:118 AMR/8000
        Media Attribute (a): fmtp:118 mode-set=7
        Media Attribute (a): rtpmap:110 telephone-event/8000
        Media Attribute (a): fmtp:110 0-15
        Media Attribute (a): ptime:20
        Media Attribute (a): sendrecv
        [Generated Call-ID: FA163E564B37-f4d-98f56700-735d25-65357ee0-9c488]

But we got an error about not available codecs and the call drops, what gives?

Both sides support AMR (Only the phone supports AMR-WB), and the Media Gateway, as the answerer, supports mode-set 7, which is supported by the UE, so we should be good?

Well, not quite:

If mode-set is specified, it MUST be abided, and frames encoded with modes outside of the subset MUST NOT be sent in any RTP payload or used in codec mode requests. If not present, all codec modes are allowed for the payload type.

RFC 4867 – RTP Payload Format for AMR and AMR-WB

Okay, I get it, the answerer (media gateway) only supports mode 7, but the UE supports all the modes, so we should be fine right?

Well, no.

Section 8.3.1 in the RFC goes on to say in the Offer-Answer Model Considerations:

The parameter [mode-set] is bi-directional, i.e., the restricted set applies to media both to be received and sent by the declaring entity. If a mode set was supplied in the offer, the answerer SHALL return the mode-set unmodified or reject the payload type. However, the answerer is free to choose a mode-set in the answer only if no mode-set was supplied in the offer for a unicast two-peer session.

And there is our problem, and why the call is getting rejected.

The Media Gateway (the answerer in this scenario) is sending back the mode-set it supports (7) but as the UE / handset (offerer) included the mode-set, the Media Gateway should either respond with the same mode set (if it supported all the requested modes) or reject it.

Instead we’re seeing the Media Gateway repond with the mode set, which it supports, which it should not do: The Media Gateway should either return the same mode-set (unmodified / unchanged) or reject it.

And boom, another ticket to another vendor…

Tales from the Trenches – Emergency Calling when Roaming

In my last post talking about the Emergency Calling Codes, I had a few comments asking about what about in roaming scenarios?

For example, an American visiting the UK, would have 911 on the Emergency Calling Codes list on their SIM card, but in the UK they dial 999 to reach emergency services.

There’s two angles to this, the first is if a roamer dials the emergency calling code of their home country, the other is if they dial the emergency calling code of the country they are in.

Let’s look at the first scenario, where the roamer dials the emergency calling code of their home country.

If our American in the UK abroad dials 911, that number is on the ECC list on the SIM, it’s still flagged as an emergency call, and just goes out with the standard urn:service:sos URN – The network never sees 911 or 999, just that it’s an SOS call that goes to the PSAP.

In this scenario, the fact the dialled number is not passed to the network is actually a positive, we get the intent that the user wants to reach emergency services, and route based on this.

But what if our American friend in need dials 999?
That’s the correct number for the end user to dial in the UK after all, but if that’s not in their ECC list on the SIM / device, it’d go through as a regular call right?

If the call does not get flagged as an emergency call on the UE this has its own set of complications and considerations:

S8-Home Routing for VoLTE means that as the UE doesn’t know this is an emergency call, the call will get routed back to the home network. This means the call doesn’t go to the E-CSCF in the visited network, and would probably just get a message saying the number they’ve dialed is unavailable, this would be exactly as if they dialed 999 at home in the US.

But we have a fix for this!
On each MME we can set a list of emergency numbers, which would allow our Britt’s phone to know on this network, what the emergency calling codes are, and route the 999 call to the local PSAP, rather than home routing it.

MME Emergency Number list Config

This information is jammed into the Emergency Number List IE in the NAS Attach Accept body.

This means our American visitor in the UK, would know about 999 from the ECC list configured in the roaming operator’s MME.

The purpose of this information element is to encode emergency number(s) for use within the country where the IE is received.

3GPP TS 24.008: 10.5.3.13 – Emergency Number List

Where this becomes more problematic is unauthenticated emergency calling.

For example, a our American visiting the UK, that is not roaming dials 999.

We’ll assume the UK and US operator don’t have a VoLTE roaming agreement because they’ve been kicking the can down the road when it comes to VoLTE roaming… This is super common scenario – last numbers I saw on this were last year with ~50 bilateral VoLTE agreements in place worldwide.

Because the phone is not attached to a local MME, the handset does not know that 999 is an emergency calling code (because it’s not on the SIM), after all, the only way it can get the Emergency Number List is from an MME, and not having been attached to an MME, means the phone does not have the ECC list for the country, so the the handset does not begin the emergency attach procedure to make the call.

Common sense prevails here, on the majority of phones and the majority of SIM profiles, codes like 112 or 911 are treated as emergency calls, but more obscure numbers, such as dialing 999 in the UK or 10111 for South African Police on a handset with US firmware, are not guaranteed to work. Generally dialing the Emergency Calling code in the home network would get you through to some emergency services (although as we talked about in the last post, this might get you routed to the wrong agency in countries where each agency has their own number).

A better way forward?

These days I don’t dial much (apart from if I’m making adjustments on the Step-by-Step exchange), when I call people I do it from contacts, hyperlinks, etc.

Emergency Dialler page in Android

There is mountains of research to suggest that asking people to remember codes and phone numbers, is a struggle. A tourist who finds themselves in Tunisia in need of assistance, is unlikely to remember that it’s 190 for an Ambulance, and 198 for Fire.

Perhaps the ECC list on a phone should populate a page of icons from the emergency page on the phone, with the universal icon for each agency, that sends to the URN for that service type?

Countries with a single PSAP could have the URNs for each service type routed to the same place, while countries with seperated PSAPs for each service type, can route accordingly.

Likewise if a country does have a centralised PSAP for all call types, knowing the type that is selected would be useful, for example if the user has pressed fire and is not responsive when the call is answered, the best unit to dispatch would probably be a fire engine.

Tales from the Trenches: The issue with Emergency Calling URNs in IMS Networks

A lot of countries have a single point of contact for emergency services; in Europe you’d call 112 in an emergency, 000 in Australia or 911 in the US. Calling this number in the country will get you the emergency services.

This means a caller can order an ambulance for smoke inhalation, and the fire brigade, in one call.

But that’s not the case in every country; many countries don’t have one number for the emergency services, they’ve got multiple; a phone number for police, a different number for fire brigade and a different number for an ambulance.

For example, in Brazil if you need the police, you call 190, while a for example, uses 193 as the emergency number for the fire department, the police can be reached at 190 or 191 depending on if it’s road policing or general, and medical emergencies are covered by 192. Other countries have similar setups.

This is all well and good if you’re in Brazil, and you call 192 for an ambulance, the phone sends a SIP INVITE with a Request URI of sip:[email protected], because we can put a rule into our E-CSCF to say if the number is 192 to route it to the answer point for ambulances – But that’s not often the case on emergency calls.

In IMS, handsets generally detect the number dialed is on the Emergency Calling Code (ECC) list from the USIM Card.

The use of the ECC list means the phone knows this is an emergency call, and this is really important. For countries that use AML this can trigger sending of the AML SMS that process, and Emergency Calls should always be allowed to be made, even without credit, a valid SIM card, or even a SIM in the phone at all.

But this comes with a cost; when a user dials 911, the phones doesn’t (generally) send a call to sip:[email protected] like it would with any other dialled number, but rather the SIP INVITE is sent to urn:service:sos which will be routed to the PSAP by the E-CSCF. When a call comes through to these URNs they’re given top priority in the network

This is all well and good in a country where it doesn’t matter which emergency service you called, because all emergency calls route to a single PSAP, but in a country with multiple numbers, it’s really important when you call and ambulance, your call doesn’t get routed to animal control.

That means the phone has to look at what emergency number you’ve dialed, and map the URN it sends the call to to match what you’ve actually requested.

Recently we’ve been helping an operator in a country with a numbering plan like this, and we’ve been finding the limits of the standards here.
So let’s start by looking at what the standards state:

IMS Emergency Calling is governed by TS 103.479 which in turn delegates to IETF RFC 5031, but for the calling number to URN translation, it’s pretty quiet.

Let’s look at what RFC 5031 allows for URNs:

  • urn:service:sos.ambulance
  • urn:service:sos.animal-control
  • urn:service:sos.fire
  • urn:service:sos.gas
  • urn:service:sos.marine
  • urn:service:sos.mountain
  • urn:service:sos.physician
  • urn:service:sos.poison
  • urn:service:sos.police

The USIM’s Emergency Calling Codes EF would be the perfect source of this data; for each emergency calling code defined, you’ve got a flag to indicate what it’s for, here’s what we’ve got available on the SIM Card:

  • Bit 1 Police
  • Bit 2 Ambulance
  • Bit 3 Fire Brigade
  • Bit 4 Marine Guard
  • Bit 5 Mountain Rescue
  • Bit 6 manually initiated eCall
  • Bit 7 automatically initiated eCall
  • Bit 8 is spare and set to “0”

So these could be mapped pretty easily you’d think, so if the call is made to an Emergency Calling Code flagged with Bit 4, the URN would go to urn:service:sos.mountain.

Alas from our research, we’ve found most OEMs send calls to the generic urn:service:sos, regardless of the dialled number and the ECC flags that are set on the SIM for that number.

One of the big chip vendors sends calls to an ECC flagged as Ambulance to urn:service:sos.fire, which is totally infuriating, and we’ve had to put a rule in our E-CSCF to handle this if the User Agent is set to one of their phones.

Is there room for improvement here? For sure! Emergency calling is super important, and time is of the essence, while animal control can probably transfer you to an ambulance, an emergency is by very nature time sensitive, and any time wasted can lead to worse outcomes.

While carrier bundles from the OEMs can handle this, the global ability to take any phone, from any country and call an emergency number is so important, that relying on a country-by-country approach here won’t suffice.

What could we do as an industry to address this?

Acknowledging that not all countries have a single point of contact for emergency service, introducing a simple mechanism in the UE SIP message to indicate what number (Emergency Calling Code) the user actually dialled would be invaluable here.

URNs are important, but knowing the dialed number when it comes to PSAP routing, is so important – This wouldn’t even need to be its own SIP header, it could just be thrown into the Contact header as another parameter.

Highly developed markets are often the first to embrace new tech (for us this means VoLTE and VoNR), but this means that these issues seen by less developed markets won’t appear until long after the standard has been set in stone, and often countries like this aren’t at the table of the standards bodies to discuss such requirements.

This easy, reasonable update to the standard, has the potential to save lives, and next time this comes up in a working group I’ll be advocating for a change.

Playing back AMR streams from Packet Captures

The other day I found myself banging my head on the table to diagnose an issue with Ringback tone on an SS7 link and the IMS.

On the IMS side, no RBT was heard, but I could see the Media Gateway was sending RTP packets to the TAS, and the TAS was sending it to the UE, but was there actual content in the RTP packets or was it just silence?

If this was PCM / G711 we’d be able to just playback in Wireshark, but alas we can’t do this for the AMR codec.

Filter the RTP stream out in Wireshark

Inside Wireshark I filtered each of the audio streams in one direction (one for the A-Party audio and one for the B-Party audio)

Then I needed to save each of the streams as a separate PCAP file (Not PCAPng).

Turn into AMR File

With the audio stream for one direction saved, we can turn it into an AMR file, using Juan Noguera (Spinlogic)’s AMR Extractor tool.

Clone the Repo from git, and then in the same directory run:

python3 pcap_parser.py -i AMR_B_Leg.pcap -o AMR_B_Leg.3ga

Playback with VLC / Audacity

I was able to play the file with VLC, and load it into Audacity to easily see that yes, the Ringback Tone was present in the AMR stream!

A look at Advanced Mobile Location SMS for Emergency Calls

Advanced Mobile Location (AML) is being rolled out by a large number of mobile network operators to provide accurate caller location to emergency services, so how does it work, what’s going on and what do you need to know?

Recently we’ve been doing a lot of work on emergency calling in IMS, and meeting requirements for NG-112 / e911, etc.

This led me to seeing my first Advanced Mobile Location (AML) SMS in the wild.

For those unfamiliar, AML is a fancy text message that contains the callers location, accuracy, etc, that is passed to emergency services when you make a call to emergency services in some countries.

It’s sent automatically by your handset (if enabled) when making a call to an emergency number, and it provides the dispatch operator with your location information, including extra metadata like the accuracy of the location information, height / floor if known, and level of confidence.

The standard is primarily driven by EENA, and, being backed by the European Union, it’s got almost universal handset support.

Google has their own version of AML called ELS, which they claim is supported on more than 99% of Android phones (I’m unclear on what this means for Harmony OS or other non-Google backed forks of Android), and Apple support for AML starts from iOS 11 onwards, meaning it’s supported on iPhones from the iPhone 5S onards,.

Call Flow

When a call is made to the PSAP based on the Emergency Calling Codes set on the SIM card or set in the OS, the handset starts collecting location information. The phone can pull this from a variety of sources, such as WiFi SSIDs visible, but the best is going to be GPS or one of it’s siblings (GLONASS / Galileo).

Once the handset has a good “lock” of a location (or if 20 seconds has passed since the call started) it bundles up all of this information the phone has, into an SMS and sends it to the PSAP as a regular old SMS.

The routing from the operator’s SMSc to the PSAP, and the routing from the PSAP to the dispatcher screen of the operator taking the call, is all up to implementation. For the most part the SMS destination is the emergency number (911 / 112) but again, this is dependent on the country.

Inside the SMS

To the user, the AML SMS is not seen, in fact, it’s actually forbidden by the standard to show in the “sent” items list in the SMS client.

On the wire, the SMS looks like any regular SMS, it can use GSM7 bit encoding as it doesn’t require any special characters.

Each attribute is a key / value pair, with semicolons (;) delineating the individual attributes, and = separating the key and the value.

Below is an example of an AML SMS body:

A"ML=1;lt=+54.76397;lg=-
0.18305;rd=50;top=20130717141935;lc=90;pm=W;si=123456789012345;ei=1234567890123456;mcc=234;mnc=30; ml=128

If you’ve got a few years of staring at Wireshark traces in Hex under your belt, then this will probably be pretty easy to get the gist of what’s going on, we’ve got the header (A”ML=1″) which denotes this is AML and the version is 1.

After that we have the latitude (lt=), longitude (lg=), radius (rd=), time of positioning (top=), level of confidence (lc=), positioning method (pm=) with G for GNSS, W for Wifi signal, C for Cell
or N for a position was not available, and so on.

AML outside the ordinary

Roaming Scenarios

If an emergency occurs inside my house, there’s a good chance I know the address, and even if I don’t know my own address, it’s probably linked to the account holder information from my telco anyway.

AML and location reporting for emergency calls is primarily relied upon in scenarios where the caller doesn’t know where they’re calling from, and a good example of this would be a call made while roaming.

If I were in a different country, there’s a much higher likelihood that I wouldn’t know my exact address, however AML does not currently work across borders.

The standard suggests disabling SMS when roaming, which is not that surprising considering the current state of SMS transport.

Without a SIM?

Without a SIM in the phone, calls can still be made to emergency services, however SMS cannot be sent.

That’s because the emergency calling standards for unauthenticated emergency calls, only cater for

This is a limitation however this could be addressed by 3GPP in future releases if there is sufficient need.

HTTPS Delivery

The standard was revised to allow HTTPS as the delivery method for AML, for example, the below POST contains the same data encoded for use in a HTTP transaction:

v=3&device_number=%2B447477593102&location_latitude=55.85732&location_longitude=-
4.26325&location_time=1476189444435&location_accuracy=10.4&location_source=GPS&location_certainty=83
&location_altitude=0.0&location_floor=5&device_model=ABC+ABC+Detente+530&device_imei=354773072099116
&device_imsi=234159176307582&device_os=AOS&cell_carrier=&cell_home_mcc=234&cell_home_mnc=15&cell_net
work_mcc=234&cell_network_mnc=15&cell_id=0213454321 

Implementation of this approach is however more complex, and leads to little benefit.

The operator must zero-rate the DNS, to allow the FQDN for this to be resolved (it resolves to a different domain in each country), and allow traffic to this endpoint even if the customer has data disabled (see what happens when your handset has PS Data Off ), or has run out of data.

Due to the EU’s stance on Net Neutrality, “Zero Rating” is a controversial topic that means most operators have limited implementation of this, so most fall back to SMS.

Other methods for sharing location of emergency calls?

In some upcoming posts we’ll look at the GMLC used for E911 Phase 2, and how the network can request the location from the handset.

Further Reading

https://eena.org/knowledge-hub/documents/aml-specifications-requirements/

VoLTE / IMS – Analysis Challenge

It’s challenge time, this time we’re going to be looking at an IMS PCAP, and answering some questions to test your IMS analysis chops!

Here’s the packet capture:

Easy Questions

  • What QCI value is used for the IMS bearer?
  • What is the registration expiry?
  • What is the E-UTRAN Cell ID the Subscriber is served by?
  • What is the AMBR of the IMS APN?

Intermediate Questions

  • Is this the first or subsequent registration?
  • What is the Integrity-Key for the registration?
  • What is the FQDN of the S-CSCF?
  • What Nonce value is used and what does it do?
  • What P-CSCF Addresses are returned?
  • What time would the UE need to re-register by in order to stay active?
  • What is the AA-Request in #476 doing?
  • Who is the(opens in a new tab)(opens in a new tab)(opens in a new tab) OEM of the handset?
  • What is the MSISDN associated with this user?

Hard Questions

  • What port is used for the ESP data?
  • Which encryption algorithm and algorithm is used?
  • How many packets are sent over the ESP tunnel to the UE?
  • Where should SIP SUBSCRIBE requests get routed?
  • What’s the model of phone?

The answers for each question are on the next page, let me know in the comments how you went, and if there’s any tricky ones!

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 40b01d74cf51bfb3c90b69b6ae7cd966d6a215d4 --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!