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.
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.
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.
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.
Every now and then when looking into a problem I have to really stop and think about how things work low down, that I haven’t thought about for a long time, and MTU is one of those things.
I faced with an LTE MTU issue recently I thought I’d go back and brush up on my MTU knowhow and do some experimenting.
Note: This is an IPv4 discussion, IPv6 does not support fragmentation.
The very, very basics
MTU is the Maximum Transmission Unit.
In practice this is the largest datagram the layer can handle, and more often than not, this is based on a physical layer constraint, in that different physical layers can only stuff so much into a frame.
“The Internet” from a consumer perspective typically has an MTU of 1500 bytes or perhaps a bit under depending on their carrier, such as 1472 bytes. SANs in data centers typically use an MTU of around 9000 bytes, Out of the box, most devices if you don’t specify, will use an MTU of 1500 bytes.
As a general rule, service providers typically try to offer an MTU as close to 1500 as possible.
Messages that are longer than the Maximum Transmission Unit need to be broken up in a process known as “Fragmenting”. Fragmenting allows large frames to be split into smaller frames to make their way across hops with a lower MTU.
All about Fragmentation
So we can break up larger packets into smaller ones by Fragmenting them, so case closed on MTU right? Sadly not.
Fragmentation leads to reduced efficiency – Fragmenting frames takes up precious CPU cycles on the router performing it, and each time a frame is broken up, additional overhead is added by the device breaking it up, and by the receiver to reassemble it.
Fragmentation can happen multiple times across a path (Multi-Stage Fragmentation). For example if a frame is sent with a length of 9000 bytes, and needs to traverse a hop with an MTU of 4000, it would need to be fragmented (broken up) into 3 frames (Frame 1 and Frame 2 would be ~4000 bytes long and frame 3 would be ~1000 bytes long). If it then needs to traverse another hop with an MTU of 1500, then the 3 fragmented frame would each need to be further fragmented, with the first frame of ~4000 bytes being split up into 3 more fragmented frames. Lost track of what just happened? Spare a thought for the routers having to to do the fragmentation and the recipient having to reassemble their packets.
Fragmented frames are reassembled by the end recipient, other devices along the transmission path don’t reassemble packets.
In the end it boils down to this trade off: The larger the packet can be, the more user data we can stuff into each one as a percentage of the overall data. We want the percentage of user data for each packet to be as high as can be. This means we want to use the largest MTU possible, without having to fragment packets.
Overhead eats into our MTU
A 1500 byte MTU that has to be encapsulated in IPsec, GTP or PPP, is no longer a 1500 byte MTU as far as the customer is concerned.
Any of these encapsulation techniques add overhead, which shrinks the MTU available to the end customer.
This means we’ve got 50 bytes of transmission / transport overhead. This will be important later on!
How do subscribers know what to use as MTU?
Typically when a subscriber buys a DSL service or HFC connection, they’ll either get a preconfigured router from their carrier, or they will be given a list of values to use that includes MTU.
LTE and 5G on the other hand tell us the value we should use.
Inside the Protocol Configuration Options in the NAS PDU, the UE requests the MTU and DNS server to be used, and is provided back from the network.
This MTU value is actually set on the MME, not the P-GW. As the MME doesn’t actually know the maximum MTU of the network, it’s up to the operator to configure this to be a value that represents the network.
Why this Matters for LTE & 5G Transmission
As we covered earlier, fragmentation is costly. If we’re fragmenting packets we are:
Wasting resources on our transmission network / core networks – as we fragment Subscriber packets it’s taking up compute resources and therefore limiting throughput
Wasting radio resources as additional overhead is introduced for fragmented packets, and additional RBs need to be scheduled to handle the fragmented packets
To test this I’ve setup a scenario in the lab, and we’ll look at the packet captures to see how the MTU is advertised, and see how big we can make our MTU on the subscriber side.
The Packet Gateway connects users of an LTE network to external networks like the Internet, by encapsulating IP packets inside GTP and forwarding them on to reach our subscriber wherever in the network they are.
So we use GTP to encapsulate user’s traffic, making it easy to carry it transparently from outside networks (Like the Internet) to the eNodeB and onto our UE / mobile phones, and more importantly redirect where the user’s traffic it’s going while keeping the same IP address.
But we need a network element to take plain old IP from external networks / Internet, and encapsulate the traffic into the GTP packets we’ll send to the subscriber.
This network element will have to do the same in reverse and decapsulate traffic coming from the subscriber to put it back onto the external networks / Internet.
That’s the role of the Packet Gateway (P-GW). The P-GW sits on the border between the outside network (An interface / reference point known as the SGi Interface) and the rest of the packet core (Serving-Gateway then onto eNodeB & UE) via the S5 Interface.
Let’s look at how the P-GW handles an incoming packet:
An IP packet comes in from the Internet destined for IP 1.2.3.4 and routed to the P-GW.
The P-GW looks up in it’s internal database what Tunnel Endpoint Identifier (TEID) IP Address 1.2.3.4 is associated with.
The P-GW encapsulates the IP packet (Layer 3 & up) into a GTP packet, adding the Tunnel Endpoint Identifier (TEID) to the GTP header.
The P-GW looks up in it’s internal database which Serving Gateway is handling traffic for that TEID.
The P-GW then sends this GTP packet containing our IP packet to the Serving Gateway.
In order to start relaying traffic to/from the S5 & SGi interfaces, the P-GW needs a set of procedures for setting up these sessions, (IP Address allocation and TEID allocation) known as bearers. This is managed using GTPv2 (aka GTPv2-Control Plane / GTPv2-C).
GTPv2-C has a set of procedures for creating these sessions, the key ones used by the P-GW are:
The Create Session Request is sent by the S-GW to the P-GW and contains the APN of the network to be setup, the IP Address to be assigned (if static) and information regarding the maximum throughput the user will be permitted to achieve.
If the P-GW was able to setup the connection as requested, a Create Session Response is sent back to the P-GW, with the IP Address for the UE to use, and the TEID (Tunnel Endpoint Identifier).
At this stage the tunnel is up and ready to go, traffic to the P-GW to the IP of the UE will be encapsulated in GTP-U packets with the TEID for this bearer, and forwarded on to the S-GW serving the user.
As our subscribers are mobile, moving between base stations / cells, the destination of the incoming GTP-U packets needs to be updated every time the subscriber moves from one cell to another.
As we covered in the last post, the Packet Gateway (P-GW) handles decapsulating and encapsulating this traffic into GTP from external networks, and vise-versa. The Packet Gateway sends the traffic onto a Serving Gateway, that forwards the GTP-U traffic onto the eNodeB serving the user.
So why not just route the traffic from the Packet Gateway directly to the eNodeB?
As our users are inherently mobile, the signalling load to keep updating the destination of the incoming GTP-U traffic to the correct eNB, would put an immense load on the P-GW. So an intermediary gateway – the Serving Gateway (S-GW), is introduced.
The S-GW handles the mobility between cells, and takes the load of the P-GW. The P-GW just hands the traffic to a S-GW and let’s the S-GW handle the mobility.
It’s worth keeping in mind that most LTE connections are not “always on”. Subscribers (UEs) go into “Idle Mode”, in which the data connection and the radio connection is essentially paused, and able to be bought back at a moments notice (this allows us to get better battery life on the UE and better frequency utilisation).
When a user enters Idle Mode, an incoming packet needs to be buffered until the Subscriber/UE can get paged and come back online. Again this function is handled by the S-GW; buffering packets until the UE comes available then forwarding them on.
A question that seems to come up often, is how to provide a static IPs to UEs on Open5GS EPC.
By default all UEs are allocated an internal IP that the server the P-GW is running on NATs out, but many users want to avoid NAT, for obvious reasons.
Open5GS has the ability to set a Static IP for each APN a subscriber has, but let’s get one thing out of the way first;
LTE is not Ethernet. No broadcast, no multicast. Each IP Address is best thought of as a single /32 network. This means you can’t have the UEs in your LTE network in the same 192.168.1.x subnet as your home network along with your laptop and printer, it’s not how it works.
So with that out of the way, let’s talk about how to do static IP address allocation in Open5GS EPC.
Assigning a Subscriber a Static IP Address
From the HSS edit the Subscriber and in the UE IPv4 or UE IPv6 address, set the static address you want to use.
You can set any UE IP Address here and it’ll get allocated to that UE.
But – there’s an issue.
The problem is not so much on the Open5GS P-GW implementation, but just how TCP/IP routing works in general.
Let’s say I assign the UE IPv4 address 1.2.3.4 to my UE. From the UE it sends a packet with the IPv4 Source address of 1.2.3.4 to anywhere on the internet, the eNB puts the packet in GTP and eventually the it gets to the P-GW which sends it out onto the internet from the source address 1.2.3.4.
The problem is that the response will never get back to me, as 1.2.3.4 is not allocated to me and will never make it back to my P-GW, so never relayed back to the UE.
For TCP traffic this means I can send the SYN with the source address of 1.2.3.4, but the SYN/ACK will be routed back to the real 1.2.3.4, and not to me, so the TCP socket will never get opened.
So while we can set a static IPs to be allocated to UEs in Open5GS, unless the traffic can be routed back to these IPs it’s not much use.
Routing
So let’s say we have assigned IP 1.2.3.4 to the UE, we’d need to put a static route on our routers to route traffic to the IP of the PGW. In my case the PGW is 10.0.1.121, so I’ll need to add a static route to get traffic destined 1.2.3.4/32 to 10.0.1.121.
In a more common case we’d assign internal IP subnets for the UE pool, and then add routes for the entire subnet to the IP of the PGW.
The S1 interface can be pretty noisy, which makes it hard to find the info you’re looking for.
So how do we find all the packets relating to a single subscriber / IMSI amidst a sea of S1 packets?
The S1 interface only contains the IMSI in certain NAS messages, so the first step in tracing a subscriber is to find the initial attach request from that subscriber containing the IMSI.
Luckily we can filter in Wireshark to find the IMSI we’re after;
Quick note – Not all IntialUEMessages will contain the IMSI – If the subscriber has already established comms with the MME it’ll instead be using a temporary identifier – M-TMSI, unless you’ve got a way to see the M-TMSI -> IMSI mapping on the MME you’ll be out of luck.
Next up let’s take a look at the contents of one of these packets,
Inside the protocolIEs is the MME_UE_S1AP_ID – This unique identifier will identify all S1 signalling for a single user.
The MME_UE_S1AP_ID is a unique identifier, assigned by the MME to identify which signaling messages are for which subscriber.
(It’s worth noting the MME_UE_S1AP_ID is only unique to the MME – If you’ve got multiple MMEs the same MME_UE_S1AP_ID could be assigned by each).
So now we have the MME_UE_S1AP_ID, we can filter all S1 messaging containing that MME_UE_S1AP_ID, we’ll use this Wireshark filter to get it:
s1ap.MME_UE_S1AP_ID == 2
Boom, there’s a all the signalling for that subscriber.
Alternatively you can just right click on the value and apply it as a filter instead of typing everything in,
Hopefully that’ll help you filter to find what you’re looking for!
In the S1-SETUP-RESPONSE and MME-CONFIGURATION-UPDATE there’s a RelativeMMECapacity (87) IE,
So what does it do?
Most eNBs support connections to multiple MMEs, for redundancy and scalability.
By returning a value from 0 to 255 the MME is able to indicate it’s available capacity to the eNB.
The eNB uses this information to determine which MME to dispatch to, for example:
MME Pool
Relative Capacity
mme001.example.com
20/255
mme002.example.com
230/255
Example MME Pooling table
The eNB with the table above would likely dispatch any incoming traffic to MME002 as MME001 has very little at capacity.
If the capacity was at 1/255 then the MME would very rarely be used.
The exact mechanism for how the MME sets it’s relative capacity is up to the MME implementer, and may vary from MME to MME, but many MMEs support setting a base capacity (for example a less powerful MME you may want to set the relative capacity to make it look more utilised).
I looked to 3GPP to find what the spec says:
On S1, no specific procedure corresponds to the NAS node selection function. The S1 interface supports the indication by the MME of its relative capacity to the eNB, in order to achieve loadbalanced MMEs within the pool area.
3GPP TS 36.410 – 5.9.2 NAS node selection function
There’s a lot of layers of signalling in the LTE / EUTRAN attach procedure, but let’s take a look at the UE attach procedure from the Network Perspective.
We won’t touch on the air interface / Uu side of things, just the EPC side of the signaling.
To make life a bit easier I’ve put different signalling messages in different coloured headings:
After a UE establishes a connection with a cell, the first step involved in the attach process is for the UE / subscriber to identify themselves and the network to authenticate them.
The TAI, EUTRAN-CGI and GUMME-ID sections all contain information about the serving network, such the tracking area code, cell global identifier and global MME ID to make up the GUTI.
The NAS part of this request contains key information about our UE and it’s capabilities, most importantly it includes the IMSI or TMSI of the subscriber, but also includes important information such as SRVCC support, different bands and RAN technologies it supports, codecs, but most importantly, the identity of the subscriber.
If this is a new subscriber to the network, the IMSI is sent as the subscriber identity, however wherever possible sending the IMSI is avoided, so if the subscriber has connected to the network recently, the M-TMSI is used instead of the IMSI, and the MME has a record of which M-TMSI to IMSI mapping it’s allocated.
Diameter: Authentication Information Request
MME to HSS
The MME does not have a subscriber database or information on the Crypto side of things, instead this functionality is offloaded to the HSS.
I’ve gone on and on about LTE UE/Subscriber authentication, so I won’t go into the details as to how this mechanism works, but the MME will send a Authentication-Information Request via Diameter to the HSS with the Username set to the Subscriber’s IMSI.
Diameter: Authentication Information Response
HSS to MME
Assuming the subscriber exists in the HSS, a Authentication-Information Answer will be sent back from the HSS via Diameter to the MME, containing the authentication vectors to send to the UE / subscriber.
Now the MME has the Authentication vectors for that UE / Subscriber it sends back a DownlinkNASTransport, Authentication response, with the NAS section populated with the RAND and AUTN values generated by the HSS in the Authentication-Information Answer.
The Subscriber / UE’s USIM looks at the AUTN value and RAND to authenticate the network, and then calculates it’s response (RES) from the RAND value to provide a RES to send back to the network.
S1AP: UplinkNASTransport, Authentication response
eNB to MME
The subscriber authenticates the network based on the sent values, and if the USIM is happy that the network identity has been verified, it generates a RES (response) value which is sent in the UplinkNASTransport, Authentication response.
The MME compares the RES sent Subscriber / UE’s USIM against the one sent by the MME in the Authentication-Information Answer (the XRES – Expected RES).
If the two match then the subscriber is authenticated.
The DownlinkNASTransport, Security mode command is then sent by the MME to the UE to activate the ciphering and integrity protection required by the network, as set in the NAS Security Algorithms section;
The MME and the UE/Subscriber are able to derive the Ciphering Key (CK) and Integrity Key (IK) from the sent crypto variables earlier, and now both know them.
S1AP: UplinkNASTransport, Security mode complete
eNB to MME
After the UE / Subscriber has derived the Ciphering Key (CK) and Integrity Key (IK) from the sent crypto variables earlier, it can put them into place as required by the NAS Security algorithms sent in the Security mode command request.
It indicates this is completed by sending the UplinkNASTransport, Security mode complete.
At this stage the authentication of the subscriber is done, and a default bearer must be established.
Diameter: Update Location Request
MME to HSS
Once the Security mode has been completed the MME signals to the HSS the Subscriber’s presence on the network and requests their Subscription-Data from the HSS.
Diameter: Update Location Answer
HSS to MME
The ULA response contains the Subscription Data used to define the data service provided to the subscriber, including the AMBR (Aggregate Maximum Bit Rate), list of valid APNs and TAU Timer.
GTP-C: Create Session Request
MME to S-GW
The MME transfers the responsibility of setting up the data bearers to the S-GW in the form of the Create Session Request.
This includes the Tunnel Endpoint Identifier (TEID) to be assigned for this UE’s PDN.
The S-GW looks at the request and forwards it onto a P-GW for IP address assignment and access to the outside world.
GTP-C: Create Session Request
S-GW to P-GW
The S-GW sends a Create Session Request to the P-GW to setup a path to the outside world.
Diameter: Credit Control Request
P-GW to PCRF
To ensure the subscriber is in a state to establish a new PDN connection (not out of credit etc), a Credit Control Request is sent to the HSS.
Diameter: Credit Control Answer
PCRF to P-GW
Assuming the Subscriber has adequate credit for this, a Credit Control Answer is sent and the P-GW and continue the PDN setup for the subscriber.
GTP-C: Create Session Response
P-GW to S-GW
The P-GW sends back a Create Session Response, containing the IP address allocated to this PDN (Framed-IP-Address).
GTP-C: Create Session Response
S-GW to MME
The S-GW slightly changes and then relays the Create Session Response back to the MME,
This message is sent to inform the eNB of the details of the PDN connection to be setup, ie AMBR, tracking area list, APN and Protocol Configuration Options,
This contains the Tunnel Endpoint Identifier (TEID) for this PDN to identify the GTP packets.
These posts focus on the use of Diameter and SIP in an IMS / VoLTE context, however these practices can be equally applied to other networks.
The Registration-Termination Request / Answer allow a Diameter Client (S-CSCF) to indicate to the HSS (Diameter Server) that it is no longer serving that user and the registration has been terminated.
Basics:
The RFC’s definition is actually pretty succinct as to the function of the Server-Assignment Request/Answer:
The Registration-Termination-Request is sent by a Diameter Multimedia server to a Diameter Multimedia client in order to request the de-registration of a user.
Reference: TS 29.229
The Registration-Termination-Request commands are sent by a S-CSCF to indicate to the Diameter server that it is no longer serving a specific subscriber, and therefore this subscriber is now unregistered.
There are a variety of reasons for this, such as PERMANENT_TERMINATION, NEW_SIP_SERVER_ASSIGNED and SIP_SERVER_CHANGE.
The Diameter Server (HSS) will typically send the Diameter Client (S-CSCF) a Registration-Termination-Answer in response to indicate it has updated it’s internal database and will no longer consider the user to be registered at that S-CSCF.
Packet Capture
I’ve included a packet capture of these Diameter Commands from my lab network which you can find below.
These posts focus on the use of Diameter and SIP in an IMS / VoLTE context, however these practices can be equally applied to other networks.
The Diameter User-Authorization-Request and User-Authorization-Answer commands are used as the first line of authorization of a user and to determine which Serving-CSCF to forward a request to.
Basics
When a SIP Proxy (I-CSCF) receives an incoming SIP REGISTER request, it sends a User-Authorization-Request to a Diameter server to confirm if the user exists on the network, and which S-CSCF to forward the request to.
When the Diameter server receives the User-Authorization-Request it looks at the User-Name (1) AVP to determine if the Domain / Realm is served by the Diameter server and the User specified exists.
Assuming the user & domain are valid, the Diameter server sends back a User-Authorization-Answer, containing a Server-Capabilities (603) AVP with the Server-Name of the S-CSCF the user will be served by.
I always find looking at the packets puts everything in context, so here’s a packet capture of both the User-Authorization-Request and the User-Authorization-Answer.
Wireshark display of User-Authorization-Request packet
Wireshark display of User-Authorization-Answer packet
First Registration
If this is the first time this Username / Domain combination (Referred to in the RFC as an AOR – Address of Record) is seen by the Diameter server in the User-Authorization-Request it will allocate a S-CSCF address for the subscriber to use from it’s pool / internal logic.
The Diameter server will store the S-CSCF it allocated to that Username / Domain combination (AoR) for subsequent requests to ensure they’re routed to the same S-CSCF.
The Diameter server indicates this is the first time it’s seen it by adding the DIAMETER_FIRST_REGISTRATION (2001) AVP to the User-Authorization-Answer.
Subsequent Registration
If the Diameter server receives another User-Authorization-Request for the same Username / Domain (AoR) it has served before, the Diameter server returns the same S-CSCF address as it did in the first User-Authorization-Answer.
It indicates this is a subsequent registration in much the same way the first registration is indicated, by adding an DIAMETER_SUBSEQUENT_REGISTRATION (2002) AVP to the User-Authorization-Answer.
User-Authorization-Type (623) AVP
An optional User-Authorization-Type (623) AVP is available to indicate the reason for the User-Authorization-Request. The possible values / reasons are:
Creating / Updating / Renewing a SIP Registration (REGISTRATION (0))
Establishing Server Capabilities & Registering (CAPABILITIES (2))
Terminating a SIP Registration (DEREGISTRATION (1))
If the User-Authorization-Type is set to DEREGISTRATION (1) then the Diameter server returns the S-CSCF address in the User-Authorization-Answer and then removes the S-SCSF address it had associated with the AoR from it’s own records.
These posts focus on the use of Diameter and SIP in an IMS / VoLTE context, however these practices can be equally applied to other networks.
The Server-Assignment-Request/Answer commands are used so a SIP Server can indicate to a Diameter server that it is serving a subscriber and pull the profile information of the subscriber.
Basics:
The RFC’s definition is actually pretty succinct as to the function of the Server-Assignment Request/Answer:
The main functions of the Diameter SAR command are to inform the Diameter server of the URI of the SIP server allocated to the user, and to store or clear it from the Diameter server.
Additionally, the Diameter client can request to download the user profile or part of it.
The Server-Assignment-Request/Answer commands are sent by a S-CSCF to indicate to the Diameter server that it is now serving a specific subscriber, (This information can then be queried using the Location-Info-Request commands) and get the subscriber’s profile, which contains the details and identities of the subscriber.
Typically upon completion of a successful SIP REGISTER dialog (Multimedia-Authentication Request), the SIP Server (S-CSCF) sends the Diameter server a Server-Assignment-Request containing the SIP Username / Domain (referred to as an Address on Record (SIP-AOR) in the RFC) and the SIP Server (S-CSCF)’s SIP-Server-URI.
The Diameter server looks at the SIP-AOR and ensures there are not currently any active SIP-Server-URIs associated with that AoR. If there are not any currently active it then stores the SIP-AOR and the SIP-Server-URI of the SIP Server (S-CSCF) serving that user & sends back a Server-Assignment-Answer.
For most request the Subscriber’s profile is also transfered to the S-SCSF in the Server-Assignment-Answer command.
SIP-Server-Assignment-Type AVP
The same Server-Assignment-Request command can be used to register, re-register, remove registration bindings and pull the user profile, through the information in the SIP-Server-Assignment-Type AVP (375),
Common values are:
NO_ASSIGNMENT (0) – Used to pull just the user profile
The Cx-User-Data profile contains the subscriber’s profile from the Diameter server in an XML formatted dataset, that is contained as part of the Server-Assignment-Answer in the Cx-User-Data AVP (606).
The profile his tells the S-CSCF what services are offered to the subscriber, such as the allowed SIP Methods (ie INVITE, MESSAGE, etc), and how to handle calls to the user when the user is not registered (ie send calls to voicemail if the user is not there).
There’s a lot to cover on the user profile which we’ll touch on in a later post.
These posts focus on the use of Diameter and SIP in an IMS / VoLTE context, however these practices can be equally applied to other networks.
The Location-Information-Request/Answer commands are used so a SIP Server query a Diameter to find which P-CSCF a Subscriber is being served by
Basics:
The RFC’s definition is actually pretty succinct as to the function of the Server-Assignment Request/Answer:
The Location-Info-Request is sent by a Diameter Multimedia client to a Diameter Multimedia server in order to request name of the server that is currently serving the user.Reference: 29.229-
The Location-Info-Request is sent by a Diameter Multimedia client to a Diameter Multimedia server in order to request name of the server that is currently serving the user.
Reference: TS 29.229
The Location-Info-Request commands is sent by an I-CSCF to the HSS to find out from the Diameter server the FQDN of the S-CSCF serving that user.
The Public-Identity AVP (601) contains the Public Identity of the user being sought.
Here you can see the I-CSCF querying the HSS via Diameter to find the S-CSCF for public identity 12722123
The Diameter server sends back the Location-Info-Response containing the Server-Name AVP (602) with the FQDN of the S-CSCF.
Packet Capture
I’ve included a packet capture of these Diameter Commands from my lab network which you can find below.
These posts focus on the use of Diameter and SIP in an IMS / VoLTE context, however these practices can be equally applied to other networks.
The Multimedia-Authentication-Request/Answer commands are used to Authenticate subscribers / UAs using a variety of mechanisms such as straight MD5 and AKAv1-MD5.
Basics:
When a SIP Server (S-CSCF) receives a SIP INVITE, SIP REGISTER or any other SIP request, it needs a way to Authenticate the Subscriber / UA who sent the request.
We’ve already looked at the Diameter User-Authorization-Request/Answer commands used to Authorize a user for access, but the Multimedia-Authentication-Request / Multimedia-Authentication-Answer it used to authenticate the user.
The SIP Server (S-CSCF) sends a Multimedia-Authentication-Request to the Diameter server, containing the Username of the user attempting to authenticate and their Public Identity.
The Diameter server generates “Authentication Vectors” – these are Precomputed cryptographic challenges to challenge the user, and the correct (“expected”) responses to the challenges. The Diameter puts these Authentication Vectors in the 3GPP-SIP-Auth-Data (612) AVP, and sends them back to the SIP server in the Multimedia-Authentication-Answer command.
The SIP server sends the Subscriber / UA a SIP 401 Unauthorized response to the initial request, containing a WWW-Authenticate header containing the challenges.
SIP 401 Response with WWW-Authenticate header populated with values from Multimedia-Auth-Answer
The Subscriber / UA sends back the initial request with the WWW-Authenticate header populated to include a response to the challenges. If the response to the challenge matches the correct (“expected”) response, then the user is authenticated.
Multimedia-Authentication-Request
Multimedia-Authentication-Answer
I always find it much easier to understand what’s going on through a packet capture, so here’s a packet capture showing the two Diameter commands,
Note: There is a variant of this process allows for stateless proxies to handle this by not storing the expected authentication values sent by the Diameter server on the SIP Proxy, but instead sending the received authentication values sent by the Subscriber/UA to the Diameter server to compare against the expected / correct values.
The Cryptography
The Cryptography for IMS Authentication relies on AKAv1-MD5 which I’ve written about before,
Essentially it’s mutual network authentication, meaning the network authenticates the subscriber, but the subscriber also authenticates the network.
One nifty feature of this interface is that you can send SMS using the MSC to switch the SMS traffic and the LTE/EUTRAN to transfer the messaging.
This means you don’t need Circuit Switched Fallback to send or receive SMS on LTE.
I assume this functionality was added to avoid the signalling load of constantly changing RAN technologies each time a subscriber sent or received an SMS, but I couldn’t find much about it’s history.
In order to get this to work you’ll essentially need the exact same setup I outlined in my CSFB example (Osmo-MSC, Osmo-STP, Osmo-HLR populated with the IMSI and MSISDN values you want to use for SMS), although you won’t actually need a GERAN / GSM radio network.
Once that’s in place you can just send SMS between subscribers,
Plus from the VTY terminal of OsmoMSC you can send SMS too:
I’ve talked about how LTE’s EUTRAN / EPC has no knowledge about voice calls or SMS and instead relies on IMS/VoLTE for these services.
Circuit Switched Fallback allows UEs to use a 2G or 3G network (Circuit Switched network) if their device isn’t connected to the IMS network to make calls as the 2G/3G network can handle the voice call or SMS routing via the Mobile Switching Center in the 2G/3G network.
However for incoming calls destined to the UE (Mobile Terminated) the MSC needs a way to keep track of which MME is serving the UE so it can get a message to the MME and the MME can relay it to the UE, to tell it to drop to a 2G or 3G network (Circuit Switched network).
The signalling between the MME (In the LTE EPC) and the MSC (In the GSM/UTRAN core) is done over the SGs interface.
While the SGs interface is primarily for managing user location state across multiple RAN types, it’s got a useful function for sending SMS over SGi, allowing users on an LTE RAN to send SMS via the MSC of the 2G/3G network (GSM/UTRAN core).
How it Works:
When a UE connects to the LTE RAN (EUTRAN) the MME signals the GSM/UMTS MSC with an SGsAP-LOCATION-UPDATE-REQUEST,
This request includes the IMSI of the subscriber that just attached and the FQDN of the MME serving that UE.
The MSC now knows that IMSI 001010000000003 is currently on LTE RAN served by MME mmec01.mmegi0002.mme.epc.mnc001.mcc001.3gppnetwork.org,
If a call or SMS comes into the MSC destined for the MSISDN of that IMSI, the MSC can page the UE on the LTE RAN to tell it to do an inter-RAN handover to GSM/UMTS.
Setting it Up
In order to get this working you’ll need OsmoMSC in place, your subscribers to exist on OsmoHLR and the LTE HSS – For example Open5GS-HSS.
Once you’ve done that the additional config on OsmoMSC is fairly simple, we just define a new SGs interface to listen on:
OsmoMSC Config:
sgs
local-port 29118
local-ip 0.0.0.0
vlr-name vlr.msc001.mnc001.mcc001.3gppnetwork.org
end
On the Open5GS side we’ve got to include the SGs info the MME config. Keep in mind the Tracking Area Code (TAC) in LTE must exist as the Location Area code (LAC) in GSM, here’s an extract of the MME section of YAML config in the Open5GS MME config:
The EUTRAN will need to advertise the presence of it’s GERAN neighbours and vise-versa so the UE/terminals know what ARFCN to move to so they don’t need to scan for the presence of other RATs when performing the handover.
Setting this up will depend on your eNB / BSC and goes beyond the scope of this post.
I’ll cover setting up neighbours in a later post as it’s a big topic.
If you don’t have neighbours configured, the handover will still work but will be much slower as the UE will have to scan to find the serving cell it’s reselecting to.
For my LTE lab I got myself a BaiCells Neutrino, it operates on Band 3 (FDD ~1800Mhz) with only 24dBm of output power max and PoE powered it works well in a lab environment without needing -48vDC supply, BBUs, DUs feeders and antennas.
Setup can be done via TR-069 or via BaiCells management server, for smaller setups the web UI makes setup pretty easy,
Logging in with admin/admin to the web interface:
We’ll select Quick Settings, and load in our MME IP address, PLMN (MCC & MNC), Tracking Area Code, Cell ID and Absolute Radio Frequency No.
Once that’s done we’ll set our Sync settings to use GPS / GNSS (I’ve attached an external GPS Antenna purchased cheaply online).
Finally we’ll set the power levels, my RF blocking setup is quite small so I don’t want excess power messing around with it, so I’ve dialed the power right back:
And that’s it, it’ll now connect to my MME on 10.0.1.133 port 36412 on SCTP.
The PLMN Identifier is used to identify the radio networks in use, it’s made up of the MCC – Mobile Country Code and MNC – Mobile Network Code.
But sadly it’s not as simple as just concatenating MCC and MNC like in the IMSI, there’s a bit more to it.
In the example above the Tracking Area Identity includes the PLMN Identity, and Wireshark has been kind enough to split it out into MCC and MNC, but how does it get that from the value 12f410?
This one took me longer to work out than I’d like to admit, and saw me looking through the GSM spec, but here goes:
PLMN Contents: Mobile Country Code (MCC) followed by the Mobile Network Code (MNC). Coding: according to TS GSM 04.08 [14].
If storage for fewer than the maximum possible number n is required, the excess bytes shall be set to ‘FF’. For instance, using 246 for the MCC and 81 for the MNC and if this is the first and only PLMN, the contents reads as follows: Bytes 1-3: ’42’ ‘F6′ ’18’ Bytes 4-6: ‘FF’ ‘FF’ ‘FF’ etc.
TS GSM 04.08 [14].
Making sense to you now? Me neither.
Here’s the Python code I wrote to encode MCC and MNCs to PLMN Identifiers and to decode PLMN into MCC and MNC, and then we’ll talk about what’s happening:
In the above example I take MCC 505 (Australia) and MCC 93 and generate the PLMN ID 05f539.
The first step in decoding is to take the first two bits (in our case 05 and reverse them – 50, then we take the third and fourth bits (f5) and reverse them too, and strip the letter f, now we have just 5. We join that with what we had earlier and there’s our MCC – 505.
Next we get our MNC, for this we take bytes 5 & 6 (39) and reverse them, and there’s our MNC – 93.
Together we’ve got MCC 505 and MNC 93.
The one answer I’m still looking for; why not just encode 50593? What is gained by encoding it as 05f539?
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