Category Archives: Mobile Networks

ENUM – DNS based Call Routing

DNS is commonly used for resolving domain names to IP Addresses, and is often described as being like “the phone book of the Internet”.

So what’s the phone book of phone books?

The answer, is (kind of) DNS. With the aid of E.164 number to URI mapping (ENUM), DNS can be used to resolve phone numbers into SIP URIs to route the traffic to.

So what is ENUM?

ENUM allows us to bypass the need for a central switch for routing calls to numbers, and instead, through a DNS lookup, resolve a phone number into a reachable SIP URI that is the ultimate destination for the traffic.

Imagine you want to call a company, you dial the phone number for that company, your phone does a DNS query against the phone number, which returns the SIP URI of the company’s PBX, and your phone sends the SIP INVITE directly to the company’s PBX, with no intermediary party carrying the call.

3GPP have specified ENUM as the prefered mechanism for resolving phone numbers into SIP addresses, and while it’s widespread adoption on the public Internet is still in its early days (See my post on The Sad story of ENUM in Australia) it is increasingly common in IMS networks and inside operator networks.

IETF defined RFC 6116 for “The E.164 to Uniform Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM)”, which defines how the system works.

So how does ENUM actually work?

ENUM allow us to lookup a phone number on a DNS server and find the SIP URI a server that will handle traffic for the phone number, but it’s a bit more complicated than the A or AAAA records you’d use to resolve a website, ENUM relies on NAPTR records.

Let’s look at the steps involved in taking an E.164 number and knowing where to send it.

Step 1 – Reverse the Numbers

We read phone numbers from left to right.

This is because historically the switch needs to get all the long-distance routing sorted first. The switch has to route your call to the exchange that serves that subscriber, which is what all the area codes and prefixes assigned to areas are all about (Throwback to SZU for any old Telco buffs).

For an E.164 number you’ve got a Country Code, Area Code and then the Subscriber Number. The number gets more specific as it goes along.

But getting more specific as you go along is the opposite how how DNS works, millions of domains share the .com suffix, and the unique / specific part is the bits before that.

So the first step in the ENUM process is to reverse the phone number, so let’s take phone number (03) 5550 0912, which in E.164 is +61 3 5550 0912.

As the spaces in the phone numbers are there for the humans, we’ll drop all of them and reverse the number, as DNS is more specific right-to-left, so we end up with

2.1.9.0.0.5.5.5.3.1.6

Step 2 – Add the Suffix

The ITU ENUM specifies the suffix e164.arpa be assigned for public ENUM entries. Private ENUM deployments may use their own suffix, but to make life simple I’m going to use e164.arpa as if it were public.

So we’ll append the e164.arpa domain onto our reversed and formatted E.164 phone number:

2.1.9.0.0.5.5.5.3.1.6.e164.arpa

Step 3 – Query it

Next we’ll run a Naming Authority Pointer (NAPTR) query against the domain, to get back a list of records for that number.

DNS is a big topic, and NAPTR and SRV takes up a good chunk of it, but what you need to know is that by using NAPTR we’re not limited to just a single response, we could have a weighted pool of servers handling traffic for this phone number, and be able to control load through the use of NAPTR, amongst other things.

Of course, if our phone can query the public NAPTR records, then so can anyone else, so we can just use a tool like Dig to query the record ourselves,

dig @10.0.1.252 -t naptr 2.1.9.0.0.5.5.5.3.1.6.e164.arpa

In the answers section I’ve setup this DNS server to only return a single response, with the regex SIP URI to use, in my case that’s sip:[email protected]

You’ll obviously need to replace the DNS server with your DNS server, and the query with the reversed and formatted version of the E.164 number you wish to query.

Step 4 – Send SIP traffic

After looking at the NAPTR records returned and using the weight and priority to determine which server/s to send to first, our phone forwards an INVITE to the URI returned in the NAPTR record.

How to interpret the returned results?

The first thing to keep in mind when working with ENUM is multiple records being returned is supported, and even encouraged.

NAPTR results return 7 fields, which define how it should be handled.

The host part is fairly obvious, and defines the host / DNS entry we’re talking about.

The Service defines what type of service this is. ENUM can be expanded beyond just voice, for example you may want to also return an email address or IM address as well as a SIP Address on an ENUM query, which you can do. By default voice uses the “E2U+sip” service to identify SIP URIs to route calls to, so in this context that’s what we’re interested in, but keep in mind there are other types out there,

Example ENUM query against a phone number showing other types of services (Email & Web)

The Order simply defines the order in which the rules are to be parsed. Lower numbers are processed first, if no matches then the next lowest, and so on until the highest number is reached.

The Pref is the processing preference. For load balancing 50/50 between two sites say a Melbourne and Sydney site, we’d return two results, with the same Order, and the same Pref, would see traffic split 50/50 between the two sites.
We could split this further, a Pref value of 10 for Melbourne, 10 for Sydney, 5 for Brisbane and 5 for Perth would see 33% of calls route to Melbourne, 33% of calls route to Sydney, 16.5% of calls route to Brisbane and 16.5% of calls route to Perth.
This is because we’d have a total preference value of 30, and the individual preference for each entry would work out as the fraction of the total (ie Pref 10 out of 30 = 10/30 or 33.3%).

The Flags denote the type of record we’re going to get, for most ENUM traffic this is going to be set to U, to denote a SIP URI with Regex.

The regexp field contains our SIP URI in the form of a Regular expression, which can include pattern matching and replacement. This is most commonly used to fill in the phone number into the SIP URI, for example instead of hardcoding the phone number into the response, we could use a Regular expression to fill in the requested number into the SIP URI.

!(^.*$)!sip:+1\[email protected]!

ENUM sounds great, how do I get it?

Here’s the tricky part.

If you’re looking to implement ENUM for an internal network, great, I’ll have some more posts here over the next few weeks covering off configuration of a DNS server to support ENUM lookups, and using Kamailio to lookup ENUM routes.

In terms of public ENUM, while many carriers are using ENUM inside their networks, public adoption of ENUM in most markets has been slow, for a number of reasons.

Many incumbent operators have been reluctant to embrace public ENUM as their role as an operator would be relegated to that of a Domain registrar.
Additionally, there’s real security risks involved in moving to ENUM – opening your phone system up to the world to accept inbound calls from anywhere. This could lead to DOS-style attacks of flooding phone numbers with automatically generated traffic, privacy risks and even less validation in terms of caller ID trust.

RIPE maintains the EnumData.org website listing the status of ENUM for each country / region.

Looking inside the MMS Exchange (With call flow and PCAP)

So you want to send an MMS?

We’ve covered SMS in the past, but MMS is a different kettle of fish.

Let’s look at how the call flow goes, when Bob wants to send a picture to Alice.

Before Bob sends the MMS, his phone will have to be setup with the correct settings to send MMS.
Sometimes this is done manually, for others it’s done through the Carrier provisioning SMS that preloads the settings, and for others it’s baked in based on the Android Carrier settings XML,

APN settings for Telstra in Australia for MMS

It’s made up of the APN to send MMS traffic over, the MMSC address (Multimedia Message Switching Center) and often an MMS proxy and port combination for where the traffic will actually go.

Message Flow – Bob to MMSC (Mobile Originated MMS)

Bob opens his phone, creates a new message to Alice, selects the picture (or other multimedia filetype) to send to her and hits the send button.

For starters, MMS has a file size limit, like MTU it’s not advertised, so you don’t know if you’ve hit it, so rather like MTU is a “lowest has the highest success of getting through” rule. So Bob’s phone will most likely scale the image down to fit inside 300K.

Next Bob’s phone knows it has an MMS to send, for this is opens up a new bearer on the MMS APN, typically called MMS, but configured in the phone by Bob.

Why use a separate APN for sending 300K of MMS traffic?
Once upon a time mobile data was expensive.
By having a separate APN just for MMS traffic (An APN that could do nothing except send / receive MMS) allowed easier billing / tariffing of data, as MMS traffic was sent over a APN which was unmetered.

After the bearer is setup on the MMS APN, Bob’s phone begins crafting a HTTP 1.1 Post to be sent to the MMSC.
The content type of this request will be application/vnd.wap.mms-message and the body of the HTTP post will be made up of MMS Message Encapsulation, with the body containing the picture he wants to send to Alice.

Note: Historically Wireless Session Protocol (WSP) was used in lieu of HTTP. These clients would now need a WAP gateway to translate into HTTP.

This HTTP Post is then sent to the MMSC Address, or, if present, the MMSC Proxy address.
This traffic is sent over the MMS APN that we just brought up.

HTTP POST Headers for the MO MMS Message

MMS Message Encapsulation from MO MMS Message

The MMSC receives this information, and then, if all was successful, responds with a 200 OK,

200 OK response to MO MMS Message

So now the MMSC has the information from Bob, let’s flip over to Alice.

Message Flow – MMSC to Alice (Mobile Terminated MMS)

For the purposes of simplicity, we’re going to rule out the MMSC from doing clever things like converting the media, accepting email (SMPP) as MMS, etc, etc. Instead we’re going to assume Alice and Bob are on the same Network, and our MMSC is just doing store-and-forward.

The MMSC will look at the To address in the MMS Message Encapsulation of the request Bob sent, to determine that this message is destined for Alice.

The MMSC will load the media content (photo) sent by Bob destined for Alice and serve it via HTTP. The MMSC generates a random URL to serve it this particular file on, with each MMS the MMSC handles being assigned a random URL containing the media content.

Next the MMSC will need to tell Alice’s phone, that she has an MMS waiting for her. This is done by generating an SMS to send to Alice’s phone,

The user-data of this SMS is the Wireless Session Protocol with the method PUSH – Aka WAP Push.

SMS alerting the user of an MMS waiting for delivery

This specially encoded SMS is parsed by the Alice’s phone, which tells the her there is an MMS message waiting for her.

On some operating systems this is pulled automatically, on others, users need to select “Download” to actually get the file.

The UE then just runs an HTTP get to the address in the X-Mms-Content-Location: Header to pull the multimedia content that Bob sent.

HTTP GET from Alice’s Phone / UE to retrieve MMS sent by Bob (MT-MMS)

All going well the URL is valid and Alice’s phone retrieves the message, getting a 200 OK back from the server with the message content.

HTTP Response (200 OK) for MT-MMS, sent by the MMSC to Alice’s phone with the MMS Body

So now Alice’s phone has the MMS content and renders it on the screen, Alice can see the Photo Bob sent her.

Lastly Alice’s phone sends a HTTP POST again to the MMSC, this time indicating the message status is “Retrieved”,

And to close everything off the MMSC confirms receipt of the Retrieved status with a 200 OK, and we are done.

What didn’t we cover?

So that’s a basic MMS message flow, but there’s a few parts we didn’t cover.

The overall architecture beyond just the store-and forward behaviour, charging and authentication we didn’t cover. So let’s look at each of these points.

Overall Architecture

What we just covered what what’s defined as the MM1 interface.

There’s obviously a stack of other interfaces, such as for charging, messaging between MMSC/Carriers, subscriber locating / user database, etc.

Charging

MMSCs would typically have a connection to trigger charging events / credit-control events prior to processing the message.

For online charging the Ro interface can be used, as you would for IMS charging events.

3GPP 3GPP TS 32.270 covers the charging architecture for online/offline charging for MMS.

Authentication

Unfortunately authentication was a bit of an afterthought for the MMS standard, and can be done several different ways.

The most common is to correlate the IP Address on the MMS APN against a subscriber.

Telephony binary-coded decimal (TBCD) in Python with Examples

Chances are if you’re reading this, you’re trying to work out what Telephony Binary-Coded Decimal encoding is. I got you.

Again I found myself staring at encoding trying to guess how it worked, reading references that looped into other references, in this case I was encoding MSISDN AVPs in Diameter.

How to Encode a number using Telephony Binary-Coded Decimal encoding?

First, Group all the numbers into pairs, and reverse each pair.

So a phone number of 123456, becomes:

214365

Because 1 & 2 are swapped to become 21, 3 & 4 are swapped to become 34, 5 & 6 become 65, that’s how we get that result.

TBCD Encoding of numbers with an Odd Length?

If we’ve got an odd-number of digits, we add an F on the end and still flip the digits,

For example 789, we add the F to the end to pad it to an even length, and then flip each pair of digits, so it becomes:

87F9

That’s the abbreviated version of it. If you’re only encoding numbers that’s all you’ll need to know.

Detail Overload

Because the numbers 0-9 can be encoded using only 4 bits, the need for a whole 8 bit byte to store this information is considered excessive.

For example 1 represented as a binary 8-bit byte would be 00000001, while 9 would be 00001001, so even with our largest number, the first 4 bits would always going to be 0000 – we’d only use half the available space.

So TBCD encoding stores two numbers in each Byte (1 number in the first 4 bits, one number in the second 4 bits).

To go back to our previous example, 1 represented as a binary 4-bit word would be 0001, while 9 would be 1001. These are then swapped and concatenated, so the number 19 becomes 1001 0001 which is hex 0x91.

Let’s do another example, 82, so 8 represented as a 4-bit word is 1000 and 2 as a 4-bit word is 0010. We then swap the order and concatenate to get 00101000 which is hex 0x28 from our inputted 82.

Final example will be a 3 digit number, 123. As we saw earlier we’ll add an F to the end for padding, and then encode as we would any other number,

F is encoded as 1111.

1 becomes 0001, 2 becomes 0010, 3 becomes 0011 and F becomes 1111. Reverse each pair and concatenate 00100001 11110011 or hex 0x21 0xF3.

Special Symbols (#, * and friends)

Because TBCD Encoding was designed for use in Telephony networks, the # and * symbols are also present, as they are on a telephone keypad.

Astute readers may have noticed that so far we’ve covered 0-9 and F, which still doesn’t use all the available space in the 4 bit area.

The extended DTMF keys of A, B & C are also valid in TBCD (The D key was sacrificed to get the F in).

Symbol4 Bit Word
*1 0 1 0
#1 0 1 1
a1 1 0 0
b1 1 0 1
c1 1 1 0

So let’s run through some more examples,

*21 is an odd length, so we’ll slap an F on the end (*21F), and then encoded each pair of values into bytes, so * becomes 1010, 2 becomes 0010. Swap them and concatenate for our first byte of 00101010 (Hex 0x2A). F our second byte 1F, 1 becomes 0001 and F becomes 1111. Swap and concatenate to get 11110001 (Hex 0xF1). So *21 becomes 0x2A 0xF1.

And as promised, some Python code from PyHSS that does it for you:

    def TBCD_special_chars(self, input):
        if input == "*":
            return "1010"
        elif input == "#":
            return "1011"
        elif input == "a":
            return "1100"
        elif input == "b":
            return "1101"
        elif input == "c":
            return "1100"      
        else:
            print("input " + str(input) + " is not a special char, converting to bin ")
            return ("{:04b}".format(int(input)))


    def TBCD_encode(self, input):
        print("TBCD_encode input value is " + str(input))
        offset = 0
        output = ''
        matches = ['*', '#', 'a', 'b', 'c']
        while offset < len(input):
            if len(input[offset:offset+2]) == 2:
                bit = input[offset:offset+2]    #Get two digits at a time
                bit = bit[::-1]                 #Reverse them
                #Check if *, #, a, b or c
                if any(x in bit for x in matches):
                    new_bit = ''
                    new_bit = new_bit + str(TBCD_special_chars(bit[0]))
                    new_bit = new_bit + str(TBCD_special_chars(bit[1]))    
                    bit = str(int(new_bit, 2))
                output = output + bit
                offset = offset + 2
            else:
                bit = "f" + str(input[offset:offset+2])
                output = output + bit
                print("TBCD_encode output value is " + str(output))
                return output
    

    def TBCD_decode(self, input):
        print("TBCD_decode Input value is " + str(input))
        offset = 0
        output = ''
        while offset < len(input):
            if "f" not in input[offset:offset+2]:
                bit = input[offset:offset+2]    #Get two digits at a time
                bit = bit[::-1]                 #Reverse them
                output = output + bit
                offset = offset + 2
            else:   #If f in bit strip it
                bit = input[offset:offset+2]
                output = output + bit[1]
                print("TBCD_decode output value is " + str(output))
                return output

The PLMN Problem for Private LTE / 5G

So it’s the not to distant future and the pundits vision of private LTE and 5G Networks was proved correct, and private networks are plentiful.

But what PLMN do they use?

The PLMN (Public Land Mobile Network) ID is made up of a Mobile Country Code + Mobile Network Code. MCCs are 3 digits and MNCs are 2-3 digits. It’s how your phone knows to connect to a tower belonging to your carrier, and not one of their competitors.

For example in Australia (Mobile Country Code 505) the three operators each have their own MCC. Telstra as the first licenced Mobile Network were assigned 505/01, Optus got 505/02 and VHA / TPG got 505/03.

Each carrier was assigned a PLMN when they started operating their network. But the problem is, there’s not much space in this range.

The PLMN can be thought of as the SSID in WiFi terms, but with a restriction as to the size of the pool available for PLMNs, we’re facing an IPv4 exhaustion problem from the start if we’re facing an explosion of growth in the space.

Let’s look at some ways this could be approached.

Everyone gets a PLMN

If every private network were to be assigned a PLMN, we’d very quickly run out of space in the range. Best case you’ve got 3 digits, so only space for 1,000 networks.

In certain countries this might work, but in other areas these PLMNs may get gobbled up fast, and when they do, there’s no more. New operators will be locked out of the market.

Loaner PLMNs

Carriers already have their own PLMNs, they’ve been using for years, some kit vendors have been assigned their own as well.

If you’re buying a private network from an existing carrier, they may permit you to use their PLMN,

Or if you’re buying kit from an existing vendor you may be able to use their PLMN too.

But what happens then if you want to move to a different kit vendor or another service provider? Do you have to rebuild your towers, reconfigure your SIMs?

Are you contractually allowed to continue using the PLMN of a third party like a hardware vendor, even if you’re no longer purchasing hardware from them? What happens if they change their mind and no longer want others to use their PLMN?

Everyone uses 999 / 99

The ITU have tried to preempt this problem by reallocating 999/99 for use in Private Networks.

The problem here is if you’ve got multiple private networks in close proximity, especially if you’re using CBRS or in close proximity to other networks, you may find your devices attempting to attach to another network with the same PLMN but that isn’t part of your network,

Mobile Country or Geographical Area Codes
Note from TSB
Following the agreement on the Appendix to Recommendation ITU-T E.212 on “shared E.212 MCC 999 for internal use within a private network” at the closing plenary of ITU-T SG2 meeting of 4 to 13 July 2018, upon the advice of ITU-T Study Group 2, the Director of TSB has assigned the Mobile Country Code (MCC) “999” for internal use within a private network. 

Mobile Network Codes (MNCs) under this MCC are not subject to assignment and therefore may not be globally unique. No interaction with ITU is required for using a MNC value under this MCC for internal use within a private network. Any MNC value under this MCC used in a network has
significance only within that network. 

The MNCs under this MCC are not routable between networks. The MNCs under this MCC shall not be used for roaming. For purposes of testing and examples using this MCC, it is encouraged to use MNC value 99 or 999. MNCs under this MCC cannot be used outside of the network for which they apply. MNCs under this MCC may be 2- or 3-digit.

(Recommendation ITU-T E.212 (09/2016))

The Crystal Ball?

My bet is we’ll see the ITU allocate an MCC – or a range of MCCs – for private networks, allowing for a pool of PLMNs to use.

When deploying networks, Private network operators can try and pick something that’s not in use at the area from a pool of a few thousand options.

The major problem here is that there still won’t be an easy way to identify the operator of a particular network; the SPN is local only to the SIM and the Network Name is only present in the NAS messaging on an attach, and only after authentication.

If you’ve got a problem network, there’s no easy way to identify who’s operating it.

But as eSIMs become more prevalent and BIP / RFM on SIMs will hopefully allow operators to shift PLMNs without too much headache.

How UEs get Time in LTE

You may have noticed in the settings on your phone the time source can be set to “Network”, but what does this actually entail and how is this information transferred?

The answer is actually quite simple,

In the NAS PDU of the Downlink NAS Transport message from the MME to the UE, is the Time Zone & Time field, which contains (unsuprisingly) the Timezone and Time.

Time is provided in UTC form with the current Timezone to show the offset.

This means that in the configuration for each TAC on your MME, you have to make sure that the eNBs in that TAC have the Timezone set for the location of the cells in that TAC, which is especially important when working across timezones.

There is no parameter for the date/time when Daylight savings time may change. But as soon as a UE goes Idle and then comes out of Idle mode, it’ll be given the updated timezone information, and during handovers the network time is also provided.
This means if you were using your phone at the moment when DST begins / ends you’d only see the updated time once the UE toggles into/out of Idle mode, or when performing a tracking-area update.

Diameter Agents

Let’s take a look at each of the common Diameter agent variants in use today:

Diameter Relay Agent / Diameter Routing Agent (DRA)

This is the simplest of the Diameter agents, but also probably the most common. The Diameter Relay agent does not look at the contents of the AVPs, it just routes messages based on the Application ID or Destination realm.

A Diameter Relay Agent does not change any AVPs except routing AVPs.

DRAs are transaction aware, but not dialog aware. This means they know if the Diameter request made it to the destination, but have no tracking of getting a response.

DRAs are common as a central hub for all Diameter hub in a network. This allows for a star topology where every Diameter service connects to a central DRA (typically two DRAs for redundancy) for a central place to manage Diameter routing, instead of having to do a full-mesh topology, which would be a nightmare on larger networks.

I recently wrote about creating a simple but unstable DRA with Kamailio.

Diameter Edge Agent

A Diameter Edge Agent is a special DRA that sits on the border between two networks and acts as a gateway between them.

Imagine a roaming exchange scenario, where each operator has to expose their core Diameter servers or DRAs to all the other operators they have roaming agreements with. Like we saw with the DRA to do a full-mesh style connection arrangement would be a mess, and wouldn’t allow internal changes inside the network without significant headaches.

Instead by putting a Diameter Edge Agent at the edge of the network, the operators who wish to access our Diameter information for roaming, only need to connect to a single point, and we can change whatever we like on the inside of the network, adding and removing servers, without having to update our roaming information (IR 21).

We can also strictly enforce security policies on rate limits and admission control, centrally, for all connections in from other operators.

Diameter Proxy Agent

The Diameter Proxy Agent does everything a DRA does, and more!

The Diameter Proxy Agent is application aware, meaning it can decode the AVPs and make decisions based upon the contents of the AVPs. It’s also able to edit / add / delete AVPs and Sub-AVPs.

These are useful for interconnect scenarios where you might need to re-write the value of an AVP, or translate a realm etc, on a Diameter request/response journey.

Diameter Translation Agent

Diameter Translation agents are used for translating between protocols, for example Diameter into MAP for GSM authentication, or into HTTP for 5G authentication.

For 5GC a new network element – the “Binding Support Function” (BSF) is introduced to translate between HTTP for 5G and Diameter for LTE, however this can be thought of as another Diameter Translation Agent.

SCTP Parameter Tuning

There’s a lot to like about SCTP. No head of line blocking, MTU issues, sequenced, acknowledged delivery of messages, not to mention Multi-Homing and message bundling.

But if you really want to get the most bang for your buck, you’ll need to tune your SCTP parameters to match the network conditions.

While tuning the parameters per-association would be time consuming, most SCTP stacks allow you to set templates for SCTP parameters, for example you would have a different set of parameters for the SCTP stacks inside your network, compared to SCTP stacks for say a roaming scenario or across microwave links.

IETF kindly provides a table with their recommended starting values for SCTP parameter tuning:

RTO.Initial3 seconds
RTO.Min1 second
RTO.Max60 seconds
Max.Burst4
RTO.Alpha1/8
RTO.Beta1/4
Valid.Cookie.Life60 seconds
Association.Max.Retrans10 attempts
Path.Max.Retrans5 attempts (per destination address)
Max.Init.Retransmits8 attempts
HB.interval30 seconds
HB.Max.Burst1
IETF – RFC4960: SCTP – Suggested Protocol Parameter Values

But by adjusting the Max Retrans and Retransmission Timeout (RTO) values, we can detect failures on the network more quickly, and reduce the number of packets we’ll loose should we have a failure.

We begin with the engineered round-trip time (RTT) – that is made up of the time it takes to traverse the link, processing time for the remote SCTP stack and time for the response to traverse the link again. For the examples below we’ll take an imaginary engineered RTT of 200ms.

RTO.min is the minimum retransmission timeout.
If this value is set too low then before the other side has had time to receive the request, process it and send a response, we’ve already retransmitted it.

This should be set to the round trip delay plus processing needed to send and acknowledge a packet plus some allowance for variability due to jitter; a value of 1.15 times the Engineered RTT is often chosen

So for us, 200 * 1.15 = 230ms RTO.min value.

RTO.max is the maximum amount of time we should wait before transmitting a request.
Typically three times the Engineered RTT.

So for us, 200 * 3 = 600ms RTO.min value.

Path.Max.Retransmissions is the maximum number of retransmissions to be sent down a path before the path is considered to be failed.
For example if we loose a transmission path on a multi-homed server, how many retransmissions along that path should we send until we consider it to be down?

Values set are dependant on if you’re multi-homing or not (you can be more picky if you are) and the level of acceptable packet loss in your transmission link.

Typical values are 4 Retransmissions (per destination address) for a Single-Homed association, and 2 Retransmissions (per destination address) for a Multi-Homed association.

Association.Max.Retransmissions is the maximum number of retransmissions for an association. If a transmission link in a multi-homed SCTP scenario were to go down, we would pass the Path.Max.Retransmissions value and the SCTP stack would stop sending traffic out that path, and try another, but what if the remote side is down? In that scenario all our paths would fail, so we need another counter – Path.Max.Retransmissions to count the total number of retransmissions to an association / destination. When the Association.Max.Retransmissions is reached the association is considered down.

In practice this value would be the number of paths, multiplied by the Path.Max.Retransmissions.

IMS Routing with iFCs

SIP routing is complicated, there’s edge cases, traffic that can be switched locally and other traffic that needs to be proxied off to another Proxy or Application server. How can you define these rules and logic in a flexible way, that allows these rules to be distributed out to multiple different network elements and adjusted on a per-subscriber basis?

Enter iFCs – The Initial Filter Criteria.

iFCs are XML encoded rules to define which servers should handle traffic matching a set of rules.

Let’s look at some example rules we might want to handle through iFCs:

  • Send all SIP NOTIFY, SUBSCRIBE and PUBLISH requests to a presence server
  • Any Mobile Originated SMS to an SMSc
  • Calls to a specific destination to a MGC
  • Route any SIP INVITE requests with video codecs present to a VC bridge
  • Send calls to Subscribers who aren’t registered to a Voicemail server
  • Use 3rd party registration to alert a server that a Subscriber has registered

All of these can be defined and executed through iFCs, so let’s take a look,

iFC Structure

iFCs are encoded in XML and typically contained in the Cx-user-data AVP presented in a Cx Server Assignment Answer response.

Let’s take a look at an example iFC and then break down the details as to what we’re specifying.

<InitialFilterCriteria>
    <Priority>10</Priority>
    <TriggerPoint>
        <ConditionTypeCNF>1</ConditionTypeCNF>
        <SPT>
            <ConditionNegated>0</ConditionNegated>
            <Group>0</Group>
            <Method>MESSAGE</Method>
        </SPT>
        <SPT>
            <ConditionNegated>0</ConditionNegated>
            <Group>1</Group>
            <SessionCase>0</SessionCase>
        </SPT>
    </TriggerPoint>
    <ApplicationServer>
        <ServerName>sip:smsc.mnc001.mcc001.3gppnetwork.org:5060</ServerName>
        <DefaultHandling>0</DefaultHandling>
    </ApplicationServer>
</InitialFilterCriteria>

Each rule in an iFC is made up of a Priority, TriggerPoint and ApplicationServer.

So for starters we’ll look at the Priority tag.
The Priority tag allows us to have multiple-tiers of priority and multiple levels of matching,
For example if we had traffic matching the conditions outlined in this rule (TriggerPoint) but also matching another rule with a lower priority, the lower priority rule would take precedence.

Inside our <TriggerPoint> tag contains the specifics of the rules and how the rules will be joined / matched, which is what we’ll focus on predominantly, and is followed by the <ApplicationServer> which is where we will route the traffic to if the TriggerPoint is matched / triggered.

So let’s look a bit more about what’s going on inside the TriggerPoint.

Each TriggerPoint is made up of Service Point Trigger (SPTs) which are individual rules that are matched or not matched, that are either combined as logical AND or logical OR statements when evaluated.

By using fairly simple building blocks of SPTs we can create a complex set of rules by joining them together.

Service Point Triggers (SPTs)

Let’s take a closer look at what goes on in an SPT.
Below is a simple SPT that will match all SIP requests using the SIP MESSAGE method request type:

        <SPT>
            <ConditionNegated>0</ConditionNegated>
            <Group>0</Group>
            <Method>MESSAGE</Method>
        </SPT>

So as you may have guessed, the <Method> tag inside the SPT defines what SIP request method we’re going to match.

But Method is only one example of the matching mechanism we can use, but we can also match on other attributes, such as Request URI, SIP Header, Session Case (Mobile Originated vs Mobile Terminated) and Session Description such as SDP.

Or an example of a SPT for anything Originating from the Subscriber utilizing the <SessionCase> tag inside the SPT.

        <SPT>
            <ConditionNegated>0</ConditionNegated>
            <Group>0</Group>
            <SessionCase>0</SessionCase>
        </SPT>

Below is another SPT that’s matching any requests where the request URI is sip:[email protected] by setting the <RequestURI> tag inside the SPT:

        <SPT>
            <ConditionNegated>0</ConditionNegated>
            <Group>0</Group>
            <RequestURI>sip:[email protected]</RequestURI>
        </SPT>

We can match SIP headers, either looking for the existence of a header or the value it is set too,

        <SPT>
            <ConditionNegated>0</ConditionNegated>
            <Group>0</Group>
            <SIPHeader>
              <Header>To</Header>
              <Content>"Nick"</Content>
            </SIPHeader>
        </SPT>

Having <Header> will match if the header is present, while the optional Content tag can be used to match

In terms of the Content this is matched using Regular Expressions, but in this case, not so regular regular expressions. 3GPP selected Extended Regular Expressions (ERE) to be used (IEEE POSIX) which are similar to the de facto standard PCRE Regex, but with a few fewer parameters.

Condition Negated

The <ConditionNegated> tag inside the SPT allows us to do an inverse match.

In short it will match anything other than what is specified in the SPT.

For example if we wanted to match any SIP Methods other than MESSAGE, setting <ConditionNegated>1</ConditionNegated> would do just that, as shown below:

        <SPT>
            <ConditionNegated>1</ConditionNegated>
            <Group>0</Group>
            <Method>MESSAGE</Method>
        </SPT>

And another example of ConditionNegated in use, this time we’re matching anything where the Request URI is not sip:[email protected]:

        <SPT>
            <ConditionNegated>1</ConditionNegated>
            <Group>0</Group>
            <RequestURI>sip:[email protected]</RequestURI>
        </SPT>

Finally the <Group> tag allows us to group together a group of rules for the purpose of evaluating.
We’ll go into it more in in the below section.

ConditionTypeCNF / ConditionTypeDNF

As we touched on earlier, <TriggerPoints> contain all the SPTs, but also, very importantly, specify how they will be interpreted.

SPTs can be joined in AND or OR conditions.

For some scenarios we may want to match where METHOD is MESSAGE and RequestURI is sip:[email protected], which is different to matching where the METHOD is MESSAGE or RequestURI is sip:[email protected].

This behaviour is set by the presence of one of the ConditionTypeCNF (Conjunctive Normal Form) or ConditionTypeDNF (Disjunctive Normal Form) tags.

If each SPT has a unique number in the GroupTag and ConditionTypeCNF is set then we evaluate as AND.

If each SPT has a unique number in the GroupTag and ConditionTypeDNF is set then we evaluate as OR.

Let’s look at how the below rule is evaluated as AND as ConditionTypeCNF is set:

<InitialFilterCriteria>
    <Priority>10</Priority>
    <TriggerPoint>
        <ConditionTypeCNF>1</ConditionTypeCNF>
        <SPT>
            <ConditionNegated>0</ConditionNegated>
            <Group>0</Group>
            <Method>MESSAGE</Method>
        </SPT>
        <SPT>
            <ConditionNegated>0</ConditionNegated>
            <Group>1</Group>
            <SessionCase>0</SessionCase>
        </SPT>
    </TriggerPoint>
    <ApplicationServer>
        <ServerName>sip:smsc.mnc001.mcc001.3gppnetwork.org:5060</ServerName>
        <DefaultHandling>0</DefaultHandling>
    </ApplicationServer>
</InitialFilterCriteria>

This means we will match if the method is MESSAGE and Session Case is 0 (Mobile Originated) as each SPT is in a different Group which leads to “and” behaviour.

If we were to flip to ConditionTypeDNF each of the SPTs are evaluated as OR.

<InitialFilterCriteria>
    <Priority>10</Priority>
    <TriggerPoint>
        <ConditionTypeDNF>1</ConditionTypeDNF>
        <SPT>
            <ConditionNegated>0</ConditionNegated>
            <Group>0</Group>
            <Method>MESSAGE</Method>
        </SPT>
        <SPT>
            <ConditionNegated>0</ConditionNegated>
            <Group>1</Group>
            <SessionCase>0</SessionCase>
        </SPT>
    </TriggerPoint>
    <ApplicationServer>
        <ServerName>sip:smsc.mnc001.mcc001.3gppnetwork.org:5060</ServerName>
        <DefaultHandling>0</DefaultHandling>
    </ApplicationServer>
</InitialFilterCriteria>

This means we will match if the method is MESSAGE and Session Case is 0 (Mobile Originated).

Where this gets a little bit more complex is when we have multiple entries in the same Group tag.

Let’s say we have a trigger point made up of:

<SPT><Method>MESSAGE</Method><Group>1</Group></SPT>
<SPT><SessionCase>0</SessionCase><Group>1</Group></SPT> 

<SPT><Header>P-Some-Header</Header><Group>2</Group></SPT> 

How would this be evaluated?

If we use ConditionTypeDNF every SPT inside the same Group are matched as AND, and SPTs with distinct are matched as OR.

Let’s look at our example rule evaluated as ConditionTypeDNF:

<ConditionTypeDNF>1</ConditionTypeDNF>
  <SPT><Method>MESSAGE</Method><Group>1</Group></SPT>
  <SPT><SessionCase>0</SessionCase><Group>1</Group></SPT> 

  <SPT><Header>P-Some-Header</Header><Group>2</Group></SPT> 

This means the two entries in Group 1 are evaluated as AND – So Method is message and Session Case is 0, OR the header “P-Some-Header” is present.

Let’s do another one, this time as ConditionTypeCNF:

<ConditionTypeCNF>1</ConditionTypeCNF>
  <SPT><Method>MESSAGE</Method><Group>1</Group></SPT>
  <SPT><SessionCase>0</SessionCase><Group>1</Group></SPT> 

  <SPT><Header>P-Some-Header</Header><Group>2</Group></SPT> 

This means the two entries in Group 1 are evaluated as OR – So Method is message OR Session Case is 0, AND the header “P-Some-Header” is present.

Pre-5G Network Slicing

Network Slicing, is a new 5G Technology. Or is it?

Pre 3GPP Release 16 the capability to “Slice” a network already existed, in fact the functionality was introduced way back at the advent of GPRS, so what is so new about 5G’s Network Slicing?

Network Slice: A logical network that provides specific network capabilities and network characteristics

3GPP TS 123 501 / 3 Definitions and Abbreviations

Let’s look at the old and the new ways, of slicing up networks, pre release 16, on LTE, UMTS and GSM.

Old Ways: APN Separation

The APN or “Access Point Name” is used so the SGSN / MME knows which gateway to that subscriber’s traffic should be terminated on when setting up the session.

APN separation is used heavily by MVNOs where the MVNO operates their own P-GW / GGSN.
This allows the MNVO can handle their own rating / billing / subscriber management when it comes to data.
A network operator just needs to setup their SGSN / MME to point all requests to setup a bearer on the MVNO’s APN to the MNVO’s gateways, and presoto, it’s no longer their problem.

Later as customers wanted MPLS solutions extended over mobile (Typically LTE), MNOs were able to offer “private APNs”.
An enterprise could be allocated an APN by the MNO that would ensure traffic on that APN would be routed into the enterprise’s MPLS VRF.
The MNO handles the P-GW / GGSN side of things, adding the APN configuration onto it and ensuring the traffic on that APN is routed into the enterprise’s VRF.

Different QCI values can be assigned to each APN, to allow some to have higher priority than others, but by slicing at an APN level you lock all traffic to those QoS characteristics (Typically mobile devices only support one primary APN used for routing all traffic), and don’t have the flexibility to steer which networks which traffic from a subscriber goes to.

It’s not really practical for everyone to have their own APNs, due in part to the namespace limitations, the architecture of how this is usually done limits this, and the simple fact of everyone having to populate an APN unique to them would be a real headache.

5G replaces APNs with “DNNs” – Data Network Names, but the functionality is otherwise the same.

In Summary:
APN separation slices all traffic from a subscriber using a special APN and provide a bearer with QoS/QCI values set for that APN, but does not allow granular slicing of individual traffic flows, it’s an all-or-nothing approach and all traffic in the APN is treated equally.

The old Ways: Dedicated Bearers

Dedicated bearers allow traffic matching a set rule to be provided a lower QCI value than the default bearer. This allows certain traffic to/from a UE to use GBR or Non-GBR bearers for traffic matching the rule.

The rule itself is known as a “TFT” (Traffic Flow Template) and is made up of a 5 value Tuple consisting of IP Source, IP Destination, Source Port, Destination Port & Protocol Number. Both the UE and core network need to be aware of these TFTs, so the traffic matching the TFT can get the QCI allocated to it.

This can be done a variety of different ways, in LTE this ranges from rules defined in a PCRF or an external interface like those of an IMS network using the Rx interface to request a dedicated bearers matching the specified TFTs via the PCRF.

Unlike with 5G network slicing, dedicated bearers still traverse the same network elements, the same MME, S-GW & P-GW is used for this traffic. This means you can’t “locally break out” certain traffic.

In Summary:
Dedicated bearers allow you to treat certain traffic to/from subscribers with different precedence & priority, but the traffic still takes the same path to it’s ultimate destination.

Old Ways: MOCN

Multi-Operator Core Network (MOCN) allows multiple MNOs to share the same active (tower) infrastructure.

This means one eNodeB can broadcast more than one PLMN and server more than one mobile network.

This slicing is very coarse – it allows two operators to share the same eNodeBs, but going beyond a handful of PLMNs on one eNB isn’t practical, and the PLMN space is quite limited (1000 PLMNs per country code max).

In Summary:
MOCN allows slicing of the RAN on a very coarse level, to slice traffic from different operators/PLMNs sharing the same RAN.

Its use is focused on sharing RAN rather than slicing traffic for users.

Adding support for AMR Codec in FreeSWITCH

If you’re building IMS Networks, the AMR config is a must, but FreeSWITCH does not ship with AMR due to licencing constraints, but has all the hard work done, you just need to add the headers for AMR support and compile.

LibOpenCore has support for AMR which we build, and then with a few minor tweaks to copy the C++ header files over to the FreeSWITCH source directory, and enable support in modules.conf.

Then when building FreeSWITCH you’ve got the AMR Codec to enable you to manage IMS / VoLTE media streams from mobile devices.

Instead of copying and pasting a list of commands to do this, I’ve published a Dockerfile here you can use to build a Docker image, or on a straight Debian Buster machine if you’re working on VMs or Bare Metal if you run the commands from the Dockerfile on the VM / bare metal.

You can find the Dockerfile on my Github here,

Diameter Droplets – The Flow-Description AVP and IPFilterRules

When it comes to setting up dedicated bearers, the Flow-Description AVP is perhaps the most important,

The specially encoded string (IPFilterRule) in the FlowDescription AVP is what our P-GW (Ok, our PCEF) uses to create Traffic Flow Templates to steer certain types of traffic down Dedicated Bearers.

So let’s take a look at how we can lovingly craft an artisanal Flow-Description.

The contents of the AVP are technically not a string, but a IPFilterRule.

IPFilterRules are actually defined in the Diameter Base Protocol (IETF RFC 6733), where we can learn the basics of encoding them,

Which are in turn based loosely off the ipfw utility in BSD.

They take the format:

action dir proto from src to dst

The action is fairly simple, for all our Dedicated Bearer needs, and the Flow-Description AVP, the action is going to be permit. We’re not blocking here.

The direction (dir) in our case is either in or out, from the perspective of the UE.

Next up is the protocol number (proto), as defined by IANA, but chances are you’ll be using 17 (UDP) or 6 (TCP) in most scenarios.

The from value is followed by an IP address with an optional subnet mask in CIDR format, for example from 10.45.0.0/16 would match everything in the 10.45.0.0/16 network.
Following from you can also specify the port you want the rule to apply to, or, a range of ports,
For example to match a single port you could use 10.45.0.0/16 1234 to match anything on port 1234, but we can also specify ranges of ports like 10.45.0.0/16 0 – 4069 or even mix and match lists and single ports, like 10.45.0.0/16 5060, 1000-2000

Protip: using any is the same as 0.0.0.0/0

Like the from, the to is encoded in the same way, with either a single IP, or a subnet, and optional ports specified.

And that’s it!

Keep in mind that Flow-Descriptions are typically sent in pairs as a minimum, as you want to match the traffic into and out of the network (not just one way), but often there can be quite a few sent, in order to match all the possible traffic that needs to be matched that may be across multiple different subnets, etc.

There is an optional Options parameter that allows you to set things like to only apply the rule to open TCP sessions, fragmentation, etc, although I’ve not seen this implemented in the wild.

Example IP filter Rules

permit in 6 from 10.98.254.0/24 5061 to 10.98.0.0/24 5060
permit out 6 from 10.98.254.0/24 5060 to 10.98.0.0/24 5061

permit in 6 from any 80 to 172.16.1.1 80
permit out 6 from 172.16.1.1 80 to any 80

permit in 17 from 10.98.254.0/24 50000-60100 to 10.98.0.0/24 50000-60100
permit out 17 from 10.98.254.0/24 50000-60100 to 10.98.0.0/24 50000-60100

permit in 17 from 10.98.254.0/24 5061, 5064 to 10.98.0.0/24  5061, 5064
permit out 17 from 10.98.254.0/24 5061, 5064 to 10.98.0.0/24  5061, 5064

permit in 17 from 172.16.0.0/16 50000-60100, 5061, 5064 to 172.16.0.0/16  50000-60100, 5061, 5064
permit out 17 from 172.16.0.0/16 50000-60100, 5061, 5064 to 172.16.0.0/16  50000-60100, 5061, 5064

For more info see:

RFC 6773 – Diameter Base Protocol – IP Filter Rule

3GPP TS 29.214 section 5.3.8 Flow-Description AVP

The Surprisingly Complicated world of MO SMS in IMS/VoLTE

Since the beginning of time, SIP has used the 2xx responses to confirm all went OK.

If you thought sending an SMS in a VoLTE/IMS network would see a 2xx OK response and then that’s the end of it, you’d be wrong.

So let’s take a look into sending SMS over VoLTE/IMS networks!

So our story starts with the Subscriber sending an SMS, which generate a SIP MESSAGE.

The Content-Type of this SIP MESSAGE is set to application/vnd.3gpp.sms rather than Text, and that’s because SMS over IMS uses the Short Message Transfer Protocol (SM-TP) inherited from GSM.

The Short Message Transfer Protocol (SM-TP) (Not related to Simple Message Transfer Protocol used in Email clients) is made up of Transfer Protocol Data Units (TPDU) that contain our message information, even though we have the Destination in our SIP headers, it’s again defined in the SM-TP body.

At first this may seem like a bit of duplication, but this allows older SMS Switching Centers (SMSc) to add support for IMS networks without any major changes, just what the SM-TP payload is wrapped up in changes.

SIP MESSAGE Request Body encoded in SM-TP

So back to our SIP MESSAGE request, typed out by the Subscriber, the UE sends this a SIP MESSAGE onto our IMS Network.

The IMS network follows it’s IFCs and routing rules, and makes it to the termination points for SMS traffic – the SMSc.

The SMSc sends back either a 200 OK or a 202 Accepted, and you’d think that’s the end of it, but no.

Our Subscriber still sees “Sending” on the screen, and the SMS is not shown as sent yet.

Instead, when the SMS has been delivered or buffered, relayed, etc, the SMSc generates a new SIP request, (as in new Call-ID / Dialog) with the request type MESSAGE, addressed to the Subscriber.

The payload of this request is another application/vnd.3gpp.sms encoded request body, again, containing SM-TP encoded data.

When the UE receives this, it will then consider the message delivered.

SM-TP encoded Delivery Report

Of course things change slightly when delivery reports are enabled, but that’s another story!

5Gethernet? – Transporting Non-IP data in 5G

I wrote not too long ago about how LTE access is not liked WiFi, after a lot of confusion amongst new Open5Gs users coming to LTE for the first time and expecting it to act like a Layer 2 network.

But 5G brings a new feature that changes that;

PDU Session Type: The type of PDU Session which can be IPv4, IPv6, IPv4v6, Ethernet or Unstructured

ETSI TS 123 501 – System Architecture for the 5G System

No longer are we limited to just IP transport, meaning at long last I can transport my Token Ring traffic over 5G, or in reality, customers can extend Layer 2 networks (Ethernet) over 3GPP technologies, without resorting to overlay networking, and much more importantly, fixed line networks, typically run at Layer 2, can leverage the 5G core architecture.

How does this work?

With TFTs and the N6 interfaces relying on the 5 value tuple with IPs/Ports/Protocol #s to make decisions, transporting Ethernet or Non-IP Data over 5G networks presents a problem.

But with fixed (aka Wireline) networks being able to leverage the 5G core (“Wireline Convergence”), we need a mechanism to handle Ethernet.

For starters in the PDU Session Establishment Request the UE indicates which PDN types, historically this was IPv4/6, but now if supported by the UE, Ethernet or Unstructured are available as PDU types.

We’ll focus on Ethernet as that’s the most defined so far,

Once an Ethernet PDU session has been setup, the N6 interface looks a bit different, for starters how does it know where, or how, to route unstructured traffic?

As far as 3GPP is concerned, that’s your problem:

Regardless of addressing scheme used from the UPF to the DN, the UPF shall be able to map the address used between the UPF and the DN to the PDU Session.

5.6.10.3 Support of Unstructured PDU Session type

In short, the UPF will need to be able to make the routing decisions to support this, and that’s up to the implementer of the UPF.

In the Ethernet scenario, the UPF would need to learn the MAC addresses behind the UE, handle ARP and use this to determine which traffic to send to which UE, encapsulate it into trusty old GTP, fill in the correct TEID and then send it to the gNodeB serving that user (if they are indeed on a RAN not a fixed network).

So where does this leave QoS? Without IPs to apply with TFTs and Packet Filter Sets to, how is this handled? In short, it’s not – Only the default QoS rule exist for a PDU Session of Type Unstructured. The QoS control for Unstructured PDUs is performed at the PDU Session level, meaning you can set the QFI when the PDU session is set up, but not based on traffic through that bearer.

Does this mean 5G RAN can transport Ethernet?

Well, it remains to be seen.

The specifications don’t cover if this is just for wireline scenarios or if it can be used on RAN.

The 5G PDU Creation signaling has a field to indicate if the traffic is Ethernet, but to work over a RAN we would need UE support as well as support on the Core.

And for E-UTRAN?

For the foreseeable future we’re going to be relying on LTE/E-UTRAN as well as 5G. So if you’re mobile with a non-IP PDU, and you enter an area only served by LTE, what happens?

PDU Session types “Ethernet” and “Unstructured” are transferred to EPC as “non-IP” PDN type (when supported by UE and network).

It is assumed that if a UE supports Ethernet PDU Session type and/or Unstructured PDU Session type in 5GS it will also support non-IP PDN type in EPS.

5.17.2 Interworking with EPC

If you were not aware of support in the EPC for Non-IP PDNs, I don’t blame you – So far support the CIoT EPS optimizations were initially for Non-IP PDN type has been for NB-IoT to supporting Non-IP Data Delivery (NIDD) for lightweight LwM2M traffic.

So why is this? Well, it may have to do with WO 2017/032399 Al which is a patent held by Ericsson, regarding “COMMUNICATION OF NON-IP DATA OVER PACKET DATA NETWORKS” which may be restricting wide scale deployment of this,

Open5Gs Logo

Open5Gs Database Schema Change

As Open5Gs has introduced network slicing, which led to a change in the database used,

Alas many users had subscribers provisioned in the old DB schema and no way to migrate the SDM data between the old and new schema,

If you’ve created subscribers on the old schema, and now after the updates your Subscriber Authentication is failing, check out this tool I put together, to migrate your data over.

The Open5Gs Python library I wrote has also been updated to support the new schema.

A very unstable Diameter Routing Agent (DRA) with Kamailio

I’d been trying for some time to get Kamailio acting as a Diameter Routing Agent with mixed success, and eventually got it working, after a few changes to the codebase of the ims_diameter_server module.

It is rather unstable, in that if it fails to dispatch to a Diameter peer, the whole thing comes crumbling down, but incoming Diameter traffic is proxied off to another Diameter peer, and Kamailio even adds an extra AVP.

Having used Kamailio for so long I was really hoping I could work with Kamailio as a DRA as easily as I do for SIP traffic, but it seems the Diameter module still needs a lot more love before it’ll be stable enough and simple enough for everyone to use.

I created a branch containing the fixes I made to make it work, and with an example config for use, but use with caution. It’s a long way from being production-ready, but hopefully in time will evolve.

https://github.com/nickvsnetworking/kamailio/tree/Diameter_Fix

PyHSS Update – YAML Config Files

One feature I’m pretty excited to share is the addition of a single config file for defining how PyHSS functions,

In the past you’d set variables in the code or comment out sections to change behaviour, which, let’s face it – isn’t great.

Instead the config.yaml file defines the PLMN, transport time (TCP or SCTP), the origin host and realm.

We can also set the logging parameters, SNMP info and the database backend to be used,

HSS Parameters
 hss:
   transport: "SCTP"
   #IP Addresses to bind on (List) - For TCP only the first IP is used, for SCTP all used for Transport (Multihomed).
   bind_ip: ["10.0.1.252"]
 #Port to listen on (Same for TCP & SCTP)
   bind_port: 3868
 #Value to populate as the OriginHost in Diameter responses
   OriginHost: "hss.localdomain"
 #Value to populate as the OriginRealm in Diameter responses
   OriginRealm: "localdomain"
 #Value to populate as the Product name in Diameter responses
   ProductName: "pyHSS"
 #Your Home Mobile Country Code (Used for PLMN calcluation)
   MCC: "999"
   #Your Home Mobile Network Code (Used for PLMN calcluation)
   MNC: "99"
 #Enable GMLC / SLh Interface
   SLh_enabled: True


 logging:
   level: DEBUG
   logfiles:
     hss_logging_file: log/hss.log
     diameter_logging_file: log/diameter.log
     database_logging_file: log/db.log
   log_to_terminal: true

 database:
   mongodb:
     mongodb_server: 127.0.0.1
     mongodb_username: root
     mongodb_password: password
     mongodb_port: 27017

 Stats Parameters
 redis:
   enabled: True
   clear_stats_on_boot: False
   host: localhost
   port: 6379
 snmp:
   port: 1161
   listen_address: 127.0.0.1
MSISDN Encoding - Brought to you by the letter F

MSISDN Encoding in Diameter AVPs – Brought to you by the letter F

So this one knocked me for six the other day,

MSISDN AVP 700 / vendor ID 10415, used to advertise the subscriber’s MSISDN in signaling.

I formatted the data as an Octet String, with the MSISDN from the database and moved on my merry way.

Not so fast…

The MSISDN AVP is of type OctetString.

This AVP contains an MSISDN, in international number format as described in ITU-T Rec E.164 [8], encoded as a TBCD-string, i.e. digits from 0 through 9 are encoded 0000 to 1001;

1111 is used as a filler when there is an odd number of digits; bits 8 to 5 of octet n encode digit 2n; bits 4 to 1 of octet n encode digit 2(n-1)+1.

ETSI TS 129 329 / 6.3.2 MSISDN AVP

Come again?

In practice this means if you have an odd lengthed MSISDN value, we need to add some padding to round it out to an even-lengthed value.

This padding happens between the last and second last digit of the MSISDN (because if we added it at the start we’d break the Country Code, etc) and as MSISDNs are variable length subscriber numbers.

1111 in octet string is best known as the letter F,

Not that complicated, just kind of confusing.

PyHSS Update – SCTP Support

Pleased to announce that PyHSS now supports SCTP for transport.

If you’re not already aware SCTP is the surprisingly attractive cousin of TCP, that addresses head of line blocking and enables multi-homing,

The fantastic PySCTP library from P1sec made adding this feature a snap. If you’re looking to add SCTP to a Python project, it’s surprisingly easy,

A seperate server (hss_sctp.py) is run to handle SCTP connections, and if you’re looking for Multihoming, we got you dawg – Just edit the config file and set the bind_ip list to include each of your IPs to multi home listen on.

And the call was coming from… INSIDE THE HOUSE. A look at finding UE Locations in LTE

Opening Tirade

Ok, admittedly I haven’t actually seen “When a Stranger Calls”, or the less popular sequel “When a stranger Redials” (Ok may have made the last one up).

But the premise (as I read Wikipedia) is that the babysitter gets the call on the landline, and the police trace the call as originating from the landline.

But you can’t phone yourself, that’s not how local loops work – When the murderer goes off hook it loops the circuit, which busys it. You could apply ring current to the line I guess externally but unless our murder has a Ring generator or has setup a PBX inside the house, the call probably isn’t coming from inside the house.

On Topic – The GMLC

The GMLC (Gateway Mobile Location Centre) is a central server that’s used to locate subscribers within the network on different RATs (GSM/UMTS/LTE/NR).

The GMLC typically has interfaces to each of the radio access technologies, there is a link between the GMLC and the CS network elements (used for GSM/UMTS) such as the HLR, MSC & SGSN via Lh & Lg interfaces, and a link to the PS network elements (LTE/NR) via Diameter based SLh and SLg interfaces with the MME and HSS.

The GMLC’s tentacles run out to each of these network elements so it can query them as to a subscriber’s location,

LTE Call Flow

To find a subscriber’s location in LTE Diameter based signaling is used, to query the MME which in turn queries, the eNodeB to find the location.

But which MME to query?

The SLh Diameter interface is used to query the HSS to find out which MME is serving a particular Subscriber (identified by IMSI or MSISDN).

The LCS-Routing-Info-Request is sent by the GMLC to the HSS with the subscriber identifier, and the LCS-Routing-Info-Response is returned by the HSS to the GMLC with the details of the MME serving the subscriber.

Now we’ve got the serving MME, we can use the SLg Diameter interface to query the MME to the location of that particular subscriber.

The MME can report locations to the GMLC periodically, or the GMLC can request the MME provide a location at that point.
For the GMLC to request a subscriber’s current location a Provide-Location-Request is set by the GMLC to the MME with the subscriber’s IMSI, and the MME responds after querying the eNodeB and optionally the UE, with the location info in the Provide-Location-Response.

(I’m in the process of adding support for these interfaces to PyHSS and all going well will release some software shortly to act at a GMLC so people can use this.)

Finding the actual Location

There are a few different ways the actual location of the UE is determined,

At the most basic level, Cell Global Identity (CGI) gives the identity of the eNodeB serving a user.
If you’ve got a 3 sector site each sector typically has its own Cell Global Identity, so you can determine to a certain extent, with the known radiation pattern, bearing and location of the sector, in which direction a subscriber is. This happens on the network side and doesn’t require any input from the UE.
But if we query the UE’s signal strength, this can then be combined with existing RF models and the signal strength reported by the UE to further pinpoint the user with a bit more accuracy. (Uplink and downlink cell coverage based positioning methods)
Barometric pressure and humidity can also be reported by the base station as these factors will impact resulting signal strengths.

Timing Advance (TA) and Time of Arrival (TOA) both rely on timing signals to/from a UE to determine it’s distance from the eNodeB. If the UE is only served by a single cell this gives you a distance from the cell and potentially an angle inside which the subscriber is. This becomes far more useful with 3 or more eNodeBs in working range of the UE, where you can “triangulate” the UE’s location. This part happens on the network side with no interaction with the UE.
If the UE supports it, EUTRAN can uses Enhanced Observed Time Difference (E-OTD) positioning method, which does TOD calcuation does this in conjunction with the UE.

GPS Assisted (A-GPS) positioning gives good accuracy but requires the devices to get it’s current location using the GPS, which isn’t part of the baseband typically, so isn’t commonly implimented.

Uplink Time Difference of Arrival (UTDOA) can also be used, which is done by the network.

So why do we need to get Subscriber Locations?

The first (and most noble) use case that springs to mind is finding the location of a subscriber making a call to emergency services. Often upon calling an emergency services number the GMLC is triggered to get the subscriber’s location in case the call is cut off, battery dies, etc.

But GMLCs can also be used for lots of other purposes, marketing purposes (track a user’s location and send targeted ads), surveillance (track movements of people) and network analytics (look at subscriber movement / behavior in a specific area for capacity planning).

Different countries have different laws regulating access to the subscriber location functions.

Hack to disable Location Reporting on Mobile Networks

If you’re wondering how you can disable this functionality, you can try the below hack to ensure that your phone does not report your location.

  1. Press the power button on your phone
  2. Turn it off

In reality, no magic super stealth SIM cards, special phones or fancy firmware will prevent the GMLC from finding your location.
So far none of the “privacy” products I’ve looked at have actually done anything special at the Baseband level. Most are just snakeoil.

For as long as your device is connected to the network, the passive ways of determining location, such as Uplink Time Difference of Arrival (UTDOA) and the CGI are going to report your location.

PyHSS New Features

Thanks to some recent developments, PyHSS has had a major overhaul recently, and is getting better than ever,

Some features that are almost ready for public release are:

Config File

Instead of having everything defined all over the place a single YAML config file is used to define how the HSS should function.

SCTP Support

No longer just limited to TCP, PyHSS now supports SCTP as well for transport,

SLh Interface for Location Services

So the GMLC can query the HSS as to the serving MME of a subscriber.

Additional Database Backends (MSSQL & MySQL)

No longer limited to just MongoDB, simple functions to add additional backends too and flexible enough to meet your existing database schema.

All these features will be merged into the mainline soon, and documented even sooner. I’ll share some posts on each of these features as I go.