Nope – it doesn’t do anything useful. So why is it there?
The SUBSCRIBE method in SIP allows a SIP UAC to subscribe to events, and then get NOTIFY messages when that event happens.
In a plain SIP scenario (RFC 3261), we can imagine an IP Phone and a PBX scenario. I might have “Busy Lamp Field” aka BLF buttons on the screen of my phone, that change colour when the people I call often are themselves on calls or on DND, so I know not to transfer calls to them – This is often called the “presence” scenario as it allows us to monitor the presence of another user.
At a SIP level, this is done by sending a SUBSCRIBE to the PBX with the information about what I’m interested in being told about (State changes for specific users) and then the PBX will send NOTIFY messages when the state changes.
But in IMS you’ll see SUBSCRIBE messages every time the subscriber registers, so what are they subscribing for?
Well, you’re just subscribing to your own registration status, but your phone knows your own registration status, because it’s, well, the registration status of the phone.
So what does it achieve? Nothing.
The idea was in a fixed-mobile-convergence scenario (keeping in mind that’s one of the key goals from the 2008 IMS spec) you could have the BLF / presence functionality for fixed subscribers, but this rareley happens.
For the past few years we’ve just been sending a 200 OK to SUBSCRIBE messages to the IMS, with a super long expiry, just to avoid wasting clock cycles.
Stumbled across these the other day, while messing around with some values on our SMSc.
Setting the Data Coding Scheme to 16 with GSM7 encoding flags the SMS as “Flash message”, which means it pops up on the screen of the phone on top of whatever the user is doing.
While reading a quality telecom blog bam! There’s the flash SMS popping over whatever I was reading.
Oddly while there’s plenty of info online about Flash SMS, it does not appear in the 3GPP specifications for SMS.
Turns out they still work, move over RCS and A2P, it’s all about Flash messages!
There’s no real secret to this other than to set the Data Coding Scheme to 16, which is GSM7 with Flash class set. That’s it.
Obviously to take advantage of this you’d need to be a network operator, or have access to the network you wish to deliver to. Recently more A2P providers are filtering non vanilla SMS traffic to filter out stuff like SMS OTA message or SIM specific messages, so there’s a good chance this may not work through A2P providers.
I’ve been writing a fair bit recently about the “VoLTE Mess” – It’s something that’s been around for a long time, mostly impacting greenfield players rolling out LTE only, but now the big carriers are starting to feel it as they shut off their 2G and 3G networks, so I figured a brief history was in order to understand how we got here.
Note: I use the terms 4G or LTE interchangeably
The Introduction of LTE
LTE (4G) is more “spectrally efficient” than the technologies that came before it. In simple terms, 1 “chunk” of spectrum will get you more speed (capacity) on LTE than the same size chunk of spectrum would on 2G or 3G.
So imagine it’s 2008 and you’re the CTO of a mobile network operator. Your network is congested thanks to carrying more data traffic than it was ever designed for (the first iPhone had launched the year before) and the network is struggling under the weight of all this new data traffic. You have two options here, to build more cell sites for more density (very expensive) or buy more spectrum (extremely expensive) – Both options see you going cap in hand to the finance team and asking for eye-wateringly large amounts of capital for either option.
But then the answer to your prayers arrives in the form of 3GPP’s Release 8 specification with the introduction of LTE. Now by taking some 2G or 3G spectrum, and by using it on 4G, you can get ~5x more capacity from the same spectrum. So just by changing spectrum you own from 2G or 3G to 4G, you’ve got 5x more capacity. Hallelujah!
So you go to Nortel and buy a packet core, and Alcatel and Siemens provide 4G RAN (eNodeBs) which you selectively deploy on the cell sites that are the most congested. The finance team and the board are happy and your marketing team runs amok with claims of 4G data speeds. You’ve dodged the crisis, phew.
This is the path that all established mobile operators took; throw LTE at the congested cell sites, to cheaply and easily free up capacity, and as the natural hardware replacement cycle kicked in, or cell sites reached capacity, swap out the hardware to kit that supports LTE in addition to the 2G and 3G tech.
Circuit Switched Fallback
But it’s hard to talk about the machinations of late 2000s telecom executives, without at least mentioning Hitler.
This video below from 15 years ago is pretty obscure and fairly technical, but the crux of it it is that Hitler is livid because LTE does not have a “CS Domain” aka circuit switched voice (the way 2G and 3G had handled voice calls).
It was optional to include support for voice calls in the LTE network (Voice over LTE) when you launched LTE services. So if you already had a 2G or 3G network (CS Network) you could just keep using 2G and 3G for your voice calls, while getting that sweet capacity relief.
So our hypothetical CTO, strapped for cash and data capacity, just didn’t bother to support VoLTE when they launched LTE – Doing so would have taken more time to launch, during which time the capacity problem would become worse, so “don’t worry about VoLTE for now” was the mantra.
All the operators who still had 2G and 3G networks, opted to just “Fallback” to using the 2G / 3G network for calling. This is called “Circuit Switched Fallback” aka CSFB.
Operators loved this as they got the capacity relief provided by shifting to 4G/LTE (more capacity in the network is always good) and could all rant about how their network was the fastest and had 4G first, this however was what could be described as a “Foot gun” – Something you can shoot yourself in the foot with in the future.
Operators eventually introduce VoLTE
Time ticked on an operators built out their 4G networks, and many in the past 10 years or so have launched VoLTE in their own networks.
For phones that support it, in areas with blanket 4G coverage, they can use VoLTE for all their calls.
But that’s the sticking point right there – If the phones support it.
But if the phones don’t support it, they’re roaming or making emergency calls, there is always been the safety blanket of 2G or 3G and Circuit Switched fallback to well, fall back to.
There’s no driver for operators who plan to (or are required to) operate a 2G or 3G network for the foreseeable future, to ensure a high level of VoLTE support in their devices.
For an operator today with 2G or 3G, Voice over LTE is still optional. Many operators still rely exclusively on Circuit Switched Fallback, and there are only a handful of countries that have turned off 2G and 3G and rely solely on VoLTE.
VoLTE Handset Support
For the past 16 years phone manufacturers have been making LTE capable phones.
But that does not mean they’ve been making phones that support Voice over LTE.
But it’s never been an issue up until this point, as there’s always been a circuit switched (2G/3G) network to fall back to, so the fact that these chips may not support VoLTE was not a big problem.
Many of the cheaper chipsets that power phones simply don’t support VoLTE – These chips do support LTE for data connections but rely on Circuit Switched Fallback for voice calls. This is in part due to the increased complexity, but also because some of the technologies for VoLTE (like AMR) required intellectual property deals to licence to use, so would add to the component cost to manufacture, and in the chips game, keeping down component cost is critical.
Even for chips that do support Voice over LTE, it’s “special”. Unlike calling in 2G or 3G that worked the same for every operator, phone manufacturers require a “Carrier Bundle” for each operator, containing that specific operators’ special flavor of VoLTE, that operator uses in their network.
This is because while VoLTE is standardized (Despite some claims to the contrary) a lot of “optional” bits have existed, and different operators built networks with subtle differences in the “flavor” of their Voice over LTE (IMS) stack they used. The OEMs (Phone / Chip manufacturers) had to handle these changes in the devices they made, for in order to sell their phones through that operator.
This means I can have a phone from vendor X that works with VoLTE on Network Y, but does not support VoLTE on Network Z.
Worse still, knowing which phones are supported is a bit of a guessing game.
Most operators sell phones directly to their customer base, so buying an Network Y branded phone from Vendor X, you know it’s going to support Network Y’s VoLTE settings, but if you change carriers, who knows if it’ll still support it?
When you’ve still got a Circuit Switched network it’s not the end of the world, you’ll just use CSFB and probably not realize it, until operators go to shut down 2G / 3G networks…
IMS Profile selection on an engineering mode MTK based Android handset
Navigating the Maze of VoLTE Compatibility
Here are some simple checklist you can ask your elderly family members if they ask if their phone is VoLTE compatible:
Does the underlying chipset the phone is based on support VoLTE? (you can find this out by disassembling the phone and checking the datasheets for the components from the OEMs after signing NDAs for each)
Does the underlying chipset require a “carrier bundle” of settings to have been loaded for this operator in order to support VoLTE (See Qualcomm MBM as an example)?
What version of this list am I currently on (generally set in the factory) and does it support this operator? (You can check by decapping the ICs and dumping their NVRAM and then running it through a decompiler)
Does my phones OS (Android / iOS) require a “carrier bundle” of it’s own to enable VoLTE? Is my operator in the version of the database on the phone? (See Android’s Carrier Database for example) (You can find the answer by rooting the phone and running some privileged commands to poke around the internal file system)
Does my operator / MNO support VoLTE – Does my plan / package support VoLTE? (You can easily find the answer by visiting the store and asking questions that don’t appear on the script)
If you managed to answer yes to all of the above, congratulations! You have conditional VoLTE support on your phone, although you probably don’t have a working phone anymore.
Wait, conditional VoLTE support?
That’s right folks, VoLTE will work in some scenarios with your operator!
If you plan on traveling, well your phone may support VoLTE at home, but does the phone have VoLTE roaming enabled? Many phones support VoLTE in the home network, but resort to CSFB when roaming.
If it does support VoLTE roaming, does the network you’re visiting support VoLTE roaming? Has the roaming agreement (IRA) between the operator you’re using while traveling and your home operator been updated to include VoLTE Roaming? These IRAs (AA.12 / AA.13 docs) also indicate if the network must turn off IPsec encryption for the VoLTE traffic when roaming, which is controlled by the phone anyway.
Phew, all this talk of VoLTE roaming while traveling scares me, I think I’ll stay home in the safety of the Australian bush with all these great friendly animals around a phone that supports VoLTE on my home network.
Ah – After spending some time in the Australian bush one of our many deadly animals bit me. Time to call for help! Wait, what about emergency calls over VoLTE? Again, many phones support VoLTE for normal calls, fall back to 2G or 3G for the emergency call, so if you have one of those phones (You’ll only find out if you try to make an emergency call and it fails) and try to make an emergency call in a country without 2G or 3G, you’d better find a payphone.
Sarcasm aside, there’s no dataset or compatibility matrix here – No simple way to see if your phone will work for VoLTE on a given operator, even if the underlying chip does support VoLTE.
Operators in Australia which recently shut down their 3G network, were mandated to block devices that didn’t support VoLTE for emergency calling. They did this using an Equipment Identity Register, and blocking devices based on the Type Allocation Code, but this scattergun approach just blocked non-carrier issued devices, regardless of it they supported VoLTE or VoLTE emergency calling.
Blame Game
So who’s to blame here?
There’s no one group to blame here, the industry has created a shitty cycle here:
Standards orgs for having too many “flavors” available
Operators deploying their own “Flavors” of VoLTE then mandating OEMs / Chip manufacturers comply with their “flavor”.
OEMs / Chip manufactures respond by adding “Carrier Bundles” to account for this per-operator customization
I’ve got some ideas on a way to unscramble this egg, and it’s going to take a push from the industry.
If you’re in the industry and keen to push for a fix, get in touch!
It’s time to get a long term solution to this problem, and we as an industry need to lead the change.
Oh boy this has been a pain in the backside with IMS / VoLTE devices using TCP and how they handle the underlying TCP sockets.
A mobile phone from manufacturer A, wants every SIP dialog to be in it’s own TCP session, while a phone from manufacturer B wants a unique TCP session per transaction, while manufacturer C thinks that every SIP message should reuse the same transaction.
So an MT call to manufacturer A, who wants every SIP dialog in it’s own transaction would look something like this:
PCSCF:44738 -> UE:5060; TCP SYN UE:5060 -> PCSCF:44738; TCP SYN/ACK PCSCF:44738 -> UE:5060; TCP ACK --- TCP connection is now open to UE from P-CSCF--- --- Start of new SIP Transaction 1 & Dialog --- PCSCF:44738 -> UE:5060; TCP PSH - SIP INVITE.... UE:5060 -> PCSCF:44738; TCP ACK
--- Start of SIP Transaction 2 --- PCSCF:44738 -> UE:5060; TCP PSH - SIP BYE.... UE:5060 -> PCSCF:44738; TCP ACK, PSH - SIP 200.... --- End of SIP Transaction 2 & SIP Dialog --- PCSCF:44738 -> UE:5060; TCP FIN UE:5060 -> PCSCF:44738; TCP ACK --- End of TCP Connection ---
Where UE:5060 – is the IP & port of the UE, as advertised in the Contact: header, while PCSCF:44738 is the PCSCF IP and a random TCP port used for this connection.
But for manufacturer B, who wants a unique TCP session per transaction, they want it to look like this:
PCSCF:44738 -> UE:5060; TCP SYN UE:5060 -> PCSCF:44738; TCP SYN/ACK PCSCF:44738 -> UE:5060; TCP ACK --- TCP connection is now open to UE from P-CSCF--- --- Start of new SIP Transaction 1 & Dialog --- PCSCF:44738 -> UE:5060; TCP PSH - SIP INVITE.... UE:5060 -> PCSCF:44738; TCP ACK
--- Start of TCP Session 2 ---- PCSCF:32627 -> UE:5060; TCP SYN UE:5060 -> PCSCF:32627; TCP SYN/ACK PCSCF:32627 -> UE:5060; TCP ACK --- Start of SIP Transaction 2 --- PCSCF:32627 -> UE:5060; TCP PSH - SIP BYE.... UE:5060 -> PCSCF:32627; TCP ACK, PSH - SIP 200.... --- End of SIP Transaction 2 & SIP Dialog --- PCSCF:32627 -> UE:5060; TCP FIN UE:5060 -> PCSCF:32627; TCP ACK --- End of TCP Connection 2 ---
And then manufacturer C wants just the one TCP session to be used for everything, so they open the TCP connection when they register, and that’s all we use for everything.
Is there any logic to this? Nope, seems to be tied to the underlying chipset (Qualcomm vs Mediatek vs Unisoc) and the SIP stack used (Qualcomm, MTK, Unisoc, Samsung, Apple).
We’ve profiled devices into one of 3 behaviors, and then we tag them based on user agent as to what “persona” they demand from the network.
I can’t believe I’m still talking about VoLTE / IMS handset support and it’s almost 2025…. For context IMS was “standardized” 17 years ago.
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).
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.
I run Ubuntu on my desktop and I mess with Kamailio a lot.
Recently I was doing some work with KEMI using Python, but installing the latest versions of Kamailio from the Debian Repos for Kamailio wasn’t working.
The following packages have unmet dependencies:
kamailio-python3-modules : Depends: libpython3.11 (>= 3.11.0) but 3.11.0~rc1-1~22.04 is to be installed
Kamailio’s Python modules expect libpython3.11 or higher, but Ubuntu 22.04 repos only contain the release candidate – not the final version:
The docs describe AttributeS as a Key-Value-Store, but that’s probably selling it short – You can do some really cool stuff with AttributeS, and in this post, we’re going to learn about using AttributeS to transform stuff.
Note: Before we get started, I’d suggest copying this config file to use for testing.
Let’s look at a really basic example, where we add some data into AttributeS, match based on Account in CGrateS, and get back that data.
Well, for starters we’re calling the SetAttributeProfile endpoint, this is where we go to create / update Attribute Profiles, but in this case, because we’re hitting it for the first time with this ID, we’re creating a new entry called “ATTR_Nick_Key_Value_Example“, this will match any Contexts (more on them later) where the FilterIDs is a string, where the request Account, is equal to 1234.
Let’s run this against the CGrateS API and take a look at the result:
This tells us we matched the Attribute with the ID ATTR_Nick_Key_Value_Example, and inside Event we can see that ExampleKey was added with value ExampleValue.
Okay, you’re saying, well what was the point of that?
Well, what if as a key in the attributes, we had the password for the SIP account, which we passed to our SIP switch (Kamailio, FreeSWITCH or Asterisk for example), and used that to authenticate?
Now if the CGrateS Agent for your SIP Switch, includes the *attributes flag, and the call is coming from 1234, we’ll get back a key called “SIP_password” with the value “sosecretiputitonthewebsite”, which you can use to auth the SIP account.
We can also return multiple AttributeS, for example, we created two Attributes (ATTR_Nick_Password_Example and ATTR_Nick_Key_Value_Example) which match on the account 1234. This means we’ll get back the SIP Password from ATTR_Nick_Password_Example and the key:value we set in ATTR_Nick_Key_Value_Example:
The order can be controlled by the Weight flag in the attribute, and if you want to stop matching any other AttributeS rules after the current Attribute, you can set the Blocker=True flag when you create/update the Attribute.
Okay, I hear you saying, that’s all well and good, I can add arbitrary key/values to stuff. Here endeth the lesson right?
Well not quite, because we can add key/values, but we can also rewrite variables using AttributeS.
Let’s imagine we’ve got 3 phone numbers (DIDs) associated with an account inside CGrateS, for example’s sake let’s say we have 12340001, 12340002 and 12340003, and we want any calls from these numbers to be billed to a CGrateS account called “NickTest1234”.
Our SIP switch doesn’t need to know anything about “NickTest1234”, just the 3 DIDs it can use to call out from your SIP stack. But to do this, we’d need CGrateS to transform any events from these DIDs to replace the Account value inside CGrateS, with NickTest1234.
In the example code to go with this I’ve put together a simple for loop to add these – You can find the code on Github (link at the bottom).
So with these defined, let’s try and rate something, we’ll add a default Charger, and add an SMS balance, before simulating an SMS where the account is set to 12340003:
#Define default Charger
print(CGRateS_Obj.SendData({"method":"APIerSv1.SetChargerProfile","params":[{"Tenant":"cgrates.org","ID":"DEFAULT","FilterIDs":[],"AttributeIDs":["*none"],"Weight":0}]}))
#Add an SMS Balance
print(CGRateS_Obj.SendData({"method":"ApierV1.SetBalance","params":[{"Tenant":"cgrates.org","Account":"Nick_Test_123","BalanceType":"*sms","Categories":"*any","Balance":{"ID":"SMS_Balance_1","Value":"100","Weight":25}}],"id":13}))
import uuid
import datetime
now = datetime.datetime.now()
result = CGRateS_Obj.SendData({
"method": "CDRsV2.ProcessExternalCDR",
"params": [
{
"OriginID": str(uuid.uuid1()),
"ToR": "*sms",
"RequestType": "*pseudoprepaid",
"AnswerTime": now.strftime("%Y-%m-%d %H:%M:%S"),
"SetupTime": now.strftime("%Y-%m-%d %H:%M:%S"),
"Tenant": "cgrates.org",
#This is going to be transformed to Nick_Test_123 by Attributes
"Account": "12340003",
"Usage": "1",
}
]
})
pprint.pprint(result)
Right, so all going well, here’s what you should see in the CDRs table:
Bingo! Despite the fact the Account in the ProcessExternalCDR was set to 12340003, and had no mention of “NickTest1234”, CGrateS transformed it to NickTest1234.
How did that happen? Well, inside our cgrates.json file we have set the cdrs and chargers modules to have a link to Attributes, which means that when we call CDRs or Chargers modules via the API, these will in turn bounce the data through AttributesS for any transformations.
This means we don’t need to run AttributeSv1.ProcessEvent ourselves, when we call CDRsV2.ProcessExternalCDR, the CDRs module will call AttributeSv1.ProcessEvent for us.
We can actually see this happening, using ngrep, which as you work more with CGrateS, is a tool you’ll get very familiar with, let’s take a peek:
sudo ngrep -t -W byline port 2012 -d lo
Now if we run the CDRsV2.ProcessExternalCDR again, we’ll see the CDRs module has called Attributes for us:
Boom, there it is, same as we ran, but it’s being handled by CGrateS for us.
If you look carefully you’ll see the context in the API request is set to “*cdrs”, this means the CDRs module is calling Attributes.
When we define each of our Attributes, as we did earlier in the post, we can set what contexts they are valid in, for example we may want to apply the transformation when called by CDRs, but not other modules, you can restrict that when you define the Attribute by setting “Contexts”: [“*cdrs”].
Okay, so we’ve done some account replacement, what else can we do?
Well, let’s look at some other use cases,
Here in Australia we’ve got a few valid dialing formats, you could dial E.164 format (Numbers look like: +61212341234), 0NSN format (Numbers look like: 02 1234 1234) or NSN format (Numbers look like: 1234 1234 assuming you’re in the 03 area code yourself). If we want to define all our Destinations in E.164 format, we’ll need to to normalise the format using AttributeS, so the numbers always come as E.164.
And then under AttributeS we’ve defined a rule to replace anything matching the 0NSN regex, to strip the first digit and append a 61, to put it in E.164 format, and in SN format as the second entry.
And there you have it folks; our number format standardized.
We can combo / cascade AttributeS rules together, with the aid of the Weight and Blocker flags in the API.
Let’s imagine the 61212341234 number has been ported from Operator1 to Operator2, and the Destinations we’ve defined in CGrateS for this prefix are currently set to DST_Operator1. But because this number has been ported we should use DST_Operator2, so we charge the Operator2, as this number has been ported.
This means we don’t need to duplicate destination definitions to show this number has been ported, as this will be updated as the call gets rated, so we just assign the Attribute to each ported number.
So let’s match where the Subject of the call is 61212341234 (even though we’re going to input the Subject as 12341234), and rewrite the Destination attribute to DST_Operator2:
From the results we can see we matched two AttributeS rules, the first, ATTR_0NSN_to_E164_02_Area_Code reformatted the Subject of the call from 12341234 to 61212341234, then the updated Subject was passed through to ATTR_Ported_61212341234, which updated the Destination attribute to DST_Operator2.
If you’re having issues, make sure you have loaded the config file, are running the latest version, and if in doubt (and not on a production system), this script will clear all the data for you so you can rule out anything interfering.
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:
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.
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.
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).
I needed to have both legs of the B2BUA bridge call through FreeSWITCH using the same Call-ID (long story), and went down the rabbit hole of looking for how to do this.
A post from 15 years ago on the mailing list from Anthony Minessale said he added “sip_outgoing_call_id” variable for this, and I found the commit, but it doesn’t work – More digging shows this variable disappears somewhere in history.
But by looking at what it changed I found sip_invite_call_id does the same thing now, so if you want to make both legs use the same Call-ID here ya go:
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.
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.
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.
We’re doing more and more network automation, and something that came up as valuable to us would be to have all the IPs in HOMER SIP Capture come up as the hostnames of the VM running the service.
Luckily for us HOMER has an API for this ready to roll, and best of all, it’s Swagger based and easily documented (awesome!).
(Probably through my own failure to properly RTFM) I was struggling to work out the correct (current) way to Authenticate against the API service using a username and password.
Because the HOMER team are awesome however, the web UI for HOMER, is just an API client.
This means to look at how to log into the API, I just needed to fire up Wireshark, log into the Web UI via my browser and then flick through the packets for a real world example of how to do this.
Homer Login JSON body as seen by Wireshark
In the Login action I could see the browser posts a JSON body with the username and password to /api/v3/auth
And in return the Homer API Server responds with a 201 Created an a auth token back:
Now in order to use the API we just need to include that token in our Authorization: header then we can hit all the API endpoints we want!
For me, the goal we were setting out to achieve was to setup the aliases from our automatically populated list of hosts. So using the info above I setup a simple Python script with Requests to achieve this:
import requests
s = requests.Session()
#Login and get Token
url = 'http://homer:9080/api/v3/auth'
json_data = {"username":"admin","password":"sipcapture"}
x = s.post(url, json = json_data)
print(x.content)
token = x.json()['token']
print("Token is: " + str(token))
#Add new Alias
alias_json = {
"alias": "Blog Example",
"captureID": "0",
"id": 0,
"ip": "1.2.3.4",
"mask": 32,
"port": 5060,
"status": True
}
x = s.post('http://homer:9080/api/v3/alias', json = alias_json, headers={'Authorization': 'Bearer ' + token})
print(x.status_code)
print(x.content)
#Print all Aliases
x = s.get('http://homer:9080/api/v3/alias', headers={'Authorization': 'Bearer ' + token})
print(x.json())
And bingo we’re done, a new alias defined.
We wrapped this up in a for loop for each of the hosts / subnets we use and hooked it into our build system and away we go!
With the Homer API the world is your oyster in terms of functionality, all the features of the Web UI are exposed on the API as the Web UI just uses the API (something I wish was more common!).
Using the Swagger based API docs you can see examples of how to achieve everything you need to, and if you ever get stuck, just fire up Wireshark and do it in the Homer WebUI for an example of how the bodies should look.
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:
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 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.
I recently fell in love with the Prometheus + Grafana combo, and I’m including it in as much of my workflow as possible, so today we’ll be integrating this with another favorite – Kamailio.
Why would we want to integrate Kamailio into Prometheus + Grafana? Observability, monitoring, alerting, cool dashboards to make it look like you’re doing complicated stuff, this duo have it all!
I’m going to assume some level of familiarity with Prometheus here, and at least a basic level of understanding of Kamailio (if you’ve never worked with Kamailio before, check out my Kamailio 101 Series, then jump back here).
So what will we achieve today?
We’ll start with the simple SIP Registrar in Kamailio from this post, and we’ll add on the xhttp_prom module, and use it to expose some stats on the rate of requests, and responses sent to those requests.
So to get started we’ll need to load some extra modules, xhttp_prom module requires xhttp (If you’d like to learn the basics of xhttp there’s also a Kamailio Bytes – xHTTP Module post covering the basics) so we’ll load both.
xHTTP also has some extra requirements to load, so in the top of our config we’ll explicitly specify what ports we want to bind to, and set two parameters that control how Kamailio handles HTTP requests (otherwise you’ll not get responses for HTTP GET requests).
Then where you load all your modules we’ll load xhttp and xhttp_prom, and set the basic parameters:
loadmodule "xhttp.so"
loadmodule "xhttp_prom.so"
# Define two counters and a gauge
modparam("xhttp_prom", "xhttp_prom_stats", "all")
By setting xhttp_prom module to expose all stats, this exposes all of Kamailio’s internal stats as counters to Prometheus – This means we don’t need to define all our own counters / histograms / gauges, instead we can use the built in ones from Kamailio. Of course we can define our own custom ones, but we’ll do that in our next post.
Lastly we’ll need to add an event route to handle HTTP requests to the /metrics URL:
