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
I’ve been experimenting with Inter-RAT & Inter-Frequency handovers recetly, and had an issue where what I thought was configured on the eNB I wasn’t seeing reflected on the UEs.
I understood the Neighbouring Cell reelection parameters are broadcast in the System Information Blocks, but how could I view them?
The answer – srsUE!
I can’t get over how cool the stuff coming out of Software Radio Systems is, but being able to simulate a UE and eNB on SDR hardware is pretty awesome, and also allows you to view low layer traces the vast majority of commercial UEs will never expose to a user.
After running srsUE with the PCAP option I let it scan for networks and find mine. I didn’t actually need to authenticate with the network, just lock to the cell.
So far we’ve focused on building a plain “2G” (voice and SMS only) network, which was all consumers expected twenty years ago.
As the number of users accessing the internet through DSL, Dial Up & ISDN grew, the idea of getting this data “on the go” became more appealing. TCP/IP was becoming the dominant standard for networking, the first 802.11 WiFi spec had recently been published and demand for mobile data was growing.
There’s a catch however – TCP/IP was never designed to be mobile.
An IP address exists in a single location.
(Disclaimer: While you can “move” a subnet by advertising itself out in a different location via BGP peering relationships with other operators, it’s cumbersome, can only be done per /24 or larger, and most importantly it’s painfully slow. IPv6 has MIPv6 which attempts to fix some of these points, but that’s outside of this scope.)
GPRS addressed the mobility issue by having a single fixed point the IP Address is assigned to (the Gateway GPRS Support Node), which encapsulates IP traffic to/from a mobile user into GTP Packet (GPRS Tunnelling Protocol), like GRE or any of the other common routing encapsulation protocols, allowing the traffic to be rerouted to different destinations as the users move from being served by one BTS to another BTS.
So now we’ve got a method of encapsulating our data we’ve got to work out how to get that data out over the air.
BTS Time Slots
Way back when we were first setting up our BSC and adding our BTS(s) you will have configured timeslots for each BTS configured on your BSC.
Chances are if you’ve been following along with this tutorial, that you configured the first time slot (timeslot 0) as a CCCH+SDCCH4, meaning Common Control Channel and 4 standalone dedicated control channels, and all the subsequent timeslots (timeslot 1 – 7) as Traffic Channels (full rate) – TCH/F.
This works well if we’re only carrying voice, but to carry data we need timeslots to put the data traffic on.
For this we’ll re assign a timeslot we were using on our BSC as a voice traffic channel (TCH/F) as a PDCH – a Packet Data Channel.
This means that on the BSC your timeslot config will look something like this:
In the above example I’ve assigned two timeslots for Packet Data Channels,
The more timeslots you allocate for data, the more bandwidth available, but the fewer voice resources available.
(Most GSM networks today have few data timeslots as more recent RATs like 3G/4G are taking the data traffic, and GSM is used primarily for voice and low bandwidth M2M communications)
GPRS and EDGE
GPRS comes in two flavors, GPRS and EDGE.
GPRS (General Packet Radio Services) was the first of the two, standardised in R97, and allowed users to reach a downlink speeds of up to 171Kbps using GMSK on the air interface to encode the data.
Users quickly expected more speed, so EDGE (Enhanced Data rates for Global Evolution) was standardised, from a core perspective it was the same, but from a BTS / Air interface perspective it relied on 8PSK instead of GMSK allowed users to reach a blistering 384Kbps on the downlink.
These speeds are the theoretical maximums.
As the difference between GPRS and EDGE is encoding on the air interface, from a core perspective it’s treated the same way, however as our BTS gets all it’s brains from the BSC, we’ll need to specify if the BTS should use EDGE or GPRS it in the BSC’s BTS config.
BSC Config
On the BSC for each BTS we want to enable for packet data, we’ll need to define the parameters.
There’s two other values we’ll introduce when setting this up,
The first is NSEI – the Network Service Entity Identifier, which is the identifier of the BTS’s Packet Control Unit, like the cell identity.
The second value we’ll touch on is the BVCI – the BSSGP Virtual Connections Identifier, which is used for addressing between the BTS PCU and the SGSN.
SS7 was first introduced in the 1970s and initially was designed for large scale setting up and tearing down of calls, but due to it’s layered architecture and prominence in the industry has been used for signalling between some CS network elements in Mobile Networks, including transporting messages between the MSC and any BSCs or RNCs it’s serving.
This is going to be fairly brief and Osmocom specific, keep in mind SS7 is a giant topic so there’s a huge amount we won’t cover.
Point Codes – SS7 Addressing & Routing
Historically SS7 networks were carried over TDM links of various types, and not over TCP/IP.
A point code is a unique address associated with each network element for addressing messages between network elements, it’s function is similar to that of an IP Address you’d use in IP networks.
When STP messaging is sent it includes a Source Point Code (SPC) and Destination Point Code (DPC).
The Signalling Transfer Point
Instead of a one-to-one connection between each SS7 device and every other SS7 device, a network element called a Signaling Transfer Point (STP) is used, which acts somewhat like a router.
The STP has an internal routing table made up of the Point Codes it has connections to and some logic to know how to get to each of them.
When it receives an SS7 message, the STP looks at the Destination point code, and finds if it has a path to that point code. If it does, it forwards the SS7 message on to the destination.
Like a router, an STP doesn’t really concern itself with the upper layer protocols and what they contain – As point codes are set in the MTP3 layer that’s the only layer the STP looks at and the upper layers aren’t really “any of its business”.
Sigtran & SS7 Over IP
As the world moved towards IP enabled everything, TDM based Sigtran Networks became increasingly expensive to maintain and operate, so a IETF taskforce called SIGTRAN was created to look at moving SS7 traffic to IP.
The first layer of SS7 were dropped it primarily concerned the physical side of the network, and in the Osmocom implementation the MTP3 layer and up were put into SCTP packets and carried on the network.
Notice I said SCTP and not TCP or UDP? I’ve touched upon SCTP on this blog before, it’s as if you took the best bits of TCP without the issues like head of line blocking and added multi-homing of connections.
To establish an SS7 connection over IP the MTP3 message an SCTP socket is established from the device to the STP, and then an ASP Maintenance message is sent, followed by a Registration Request containing it’s point code, and presto, we have a connection.
The Osmo STP
The Osmocom STP acts in a very trusting manner by default,
When a device wants to connect to the STP it does so via a REG_REQ (Registration Request) containing it’s Point Code. The STP accepts the connection with a REG_RSP (Registration Response).
For as long as that connection stays up any SS7 messages destined to that point code of the device that just registered, the STP will now how to get it there.
Assuming you’ve already installed the OsmoSTP you can access it on 4239:
root@gsm-bts:/etc/osmocom# telnet localhost 4239
Trying 127.0.0.1…
Connected to localhost.
Welcome to the OsmoSTP VTY interface
OsmoSTP>
After running enable we can check the current routing table:
OsmoSTP# show cs7 instance 0 sccp users
SS7 instance 0 has no SCCP
OsmoSTP# show cs7 instance 0 ro
OsmoSTP# show cs7 instance 0 route
Routing table = system
C=Cong Q=QoS P=Prio
Destination C Q P Linkset Name Linkset Non-adj Route
0.23.1/14 0 as-rkm-1 ? ? ?
0.23.3/14 0 as-rkm-2 ? ? ?
OsmoSTP# show cs7 instance 0 as all
Routing Routing Key Cic Cic Traffic
AS Name State Context Dpc Si Opc Ssn Min Max Mode
as-rkm-1 AS_ACTIVE 1 0.23.1 override
as-rkm-2 AS_ACTIVE 2 0.23.3 override
OsmoSTP# show cs7 instance 0 asp
Effect Primary
ASP Name AS Name State Type Remote IP Addr:Rmt Port SCTP
------------ ------------ ------------- ---- ----------------------- ----------
asp-dyn-0 ? ASP_ACTIVE m3ua 127.0.0.1:52192
asp-dyn-1 ? ASP_ACTIVE m3ua 127.0.0.1:33570
Packet Capture
Below is a packet capture showing a connection from an MSC to the STP.
This is a really useful Feature that allows you to break up your S-GW (And by extension P-GW) selection based on geographical areas.
This can be used to enable Local Breakout to a S/P-GW located at the same site as the tower, but controlled by a central MME / HSS.
After updating to the latest version the configuration is pretty straightforard,
P-GW Selection based on eNB ID
# o SGW selection by eNodeB ID (either single enb_id or multiple enb_ids, decimal or hex representation)
#
selection_mode: enb_id
gtpc:
- addr: 127.0.2.3
enb_id: [9413, 0x98765]
The above config will send any traffic from eNBs with the eNB ID 9413 (encoded as an intiger) or 0x98765 (Encoded as hex int equivilent 624485) to an S-GW at 127.0.2.3.
P-GW Selection based on TAC
# SGW selection by eNodeB TAC (either single TAC or multiple TACs)
#
selection_mode: tac
gtpc:
- addr: 127.0.2.2
tac: [25000, 27000, 28000]
The above config will send any traffic from eNBs with TACs of 25000, 27000, 28000 to an S-GW at 127.0.2.2.
The Origin-State-Id AVP solves a kind of tricky problem – how do you know if a Diameter peer has restarted?
It seems like a simple problem until you think about it. One possible solution would be to add an AVP for “Recently Rebooted”, to be added on the first command queried of it from an endpoint, but what if there are multiple devices connecting to a Diameter endpoint?
The Origin-State AVP is a strikingly simple way to solve this problem. It’s a constantly incrementing counter that resets if the Diameter peer restarts.
If a client receives a Answer/Response where the Origin-State AVP is set to 10, and then the next request it’s set to 11, then the one after that is set to 12, 13, 14, etc, and then a request has the Origin-State AVP set to 5, the client can tell when it’s restarted by the fact 5 is lower than 14, the one before it.
It’s a constantly incrementing counter, that allows Diameter peers to detect if the endpoint has restarted.
Simple but effective.
You can find more about this in RFC3588 – the Diameter Base Protocol.
If you’re using BaiCells hardware you may have noticed the new eNBs and USIMs are shipping with the PLMN of MCC 314 / MNC 030.
First thing I do is change the PLMN, but I was curious as to why the change.
It seems 314 / 030 was never assigned to BaiCells to use and when someone picked this up they were forced to change it.
The MCC (Mobile Country Code) part is dictated by the country / geographic area the subscribers’ are in, as defined by ITU, whereas the MNC (Mobile Network Code) allocation is managed by the regional authority and ITU are informed as to what the allocations are and publish in their bulletins.
Well, SIM cards will have a different IMSI / PLMN, but the hardware supports Multi-Operator Core Network which allows one eNB to broadcast multiple PLMNs, so if you update your eNB it can broadcast both!
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.
You may want to connect Open5GS’ MME to a different Home Subscriber Server (HSS),
To do it we need a few bits of information:
The Domain Name of the HSS
The Realm of the HSS
The IP of the HSS
The Transport Used (TCP/SCTP)
If TLS is used
With these bits of information we can go about modifying the Open5GS MME config to talk to our different HSS.
Edit FreeDiameter Config
The config for the Open5GS MME’s Diameter peers is handled by the FreeDimaeter library,
You can find it’s config files in:
/etc/freediameter/mme.conf
We’ll start by changing the realm to match the realm of the HSS and the identity to match the identity configured as the MME peer on the HSS.
We’ll next set the ListenOn address to be a reachable IP address isntead of just a loopback address,
If you’re using TLS you’ll need to put your certificates and private key files into the TLS config,
Finally we’ll put our HSS details in the Peer Configuration;
Once all this is done we’ll need to restart our MME and you should see the Diameter Capabilities Exchange / Answer commands between the HSS and the MME if all was successful,
And that’s it! We’re connected to an external HSS.
Through the freeDiameter config file you can specify multiple ConnectPeer() entries to connect to multiple HSS (like a pool of them), and requests will be distributed evenly between them.
So SDR is all well and good, but a late night eBay purchase landed me two ipaccess NanoBTS units second hand from the US.
The hefty metal units are designed as GSM access points / picocells for indoor use, with a stable Uu / radio interface and speaking Abis over IP, it integrates nicely with Osmocom’s stack and was used by the Osmocom team as a bit of a development platform in the past.
Finding the Current IP
Because these units are second hand, first step was finding the current IP.
I ran a packet capture on the interface the units were plugged into until I saw some traffic showing their current IP.
Once you’re in the correct subnet you can use the abisip-find tool to find any units:
abisip-find
Mine showed up on a 10.97.99.15 IP, so I put my machine on the 10.97.99.x/24 subnet so I could reach them.
Changing IP Details
Once I had the current IP details it was time to change the IP details, Unit ID and OML / AbisIP IP address.
My unit came on 10.97.99.15, but I wanted it on 10.0.1.204/24 and pointed to my BSC at 10.0.1.201, so I set that using the command,
Note: I’m running version 19.12.0 which I installed from the repos due to issues with 20.4.0 (latest when I wrote this) and stability on LimeSDR.
I wrote the other day about installing SRS LTE stack,
But installing it is one thing, meeting all the requirements to use it with your SDR hardware turns out to be another whole thing all together.
srsENB is a software defined eNodeB, allowing you to use a Software Defined Radio to serve as an eNodeB, UE and a few other utilities.
SRS’ implementation of the eNB is supposed to be 3GPP R10 compliant and supports eMBMS to boot.
Meeting Dependencies
Installing prerequisites
I’m using a LimeSDR, but these instructions also for for the BladeRF. I found the frequency stability of my BladeRF X40 wasn’t great, meaning when running SRS’s eNodeB the cell wasn’t visible to my UE.
sudo apt install tree vim git g++ make cmake pkg-config python-numpy swig libi2c-dev libusb-1.0-0-dev libfftw3-dev libmbedtls-dev libboost-program-options-dev libconfig++-dev libsctp-dev gnuradio
Install SoapySDR from Source
git clone https://github.com/pothosware/SoapySDR.git pushd SoapySDR git checkout tags/soapy-sdr-0.7.2 -b soapy-sdr-0.7.2 mkdir build cd build cmake .. make sudo make install sudo ldconfig popd
Install LimeSuite
You can skip this if you’re using a BladeRF
git clone https://github.com/myriadrf/LimeSuite.git
pushd LimeSuite
#git checkout tags/v19.04.0 -b v19.04.0
mkdir builddir
cd builddir
cmake ..
make
sudo make install
sudo ldconfig
cd ../udev-rules
sudo sh ./install.sh
popd
Install BladeRF
You can skip this if using a LimeSDR
git clone https://github.com/Nuand/bladeRF.git
pushd bladeRF/host/
mkdir build
cd build/
cmake -DCMAKE_BUILD_TYPE=Release -DCMAKE_INSTALL_PREFIX=/usr/local -DINSTALL_UDEV_RULES=ON -DBLADERF_GROUP=plugdev ..
make
sudo make install
sudo ldconfig
sudo mkdir -p /etc/Nuand/bladeRF/
sudo wget https://www.nuand.com/fpga/hostedx40-latest.rbf --output-document /etc/Nuand/bladeRF/hostedx40.rbf
popd
git clone https://github.com/pothosware/SoapyBladeRF.git
pushd SoapyBladeRF
mkdir build
cd build
cmake ..
make
sudo make install
popd
Install SRS GUI
(Optional but makes life easier and has to be done prior to installing SRSLTE)
sudo apt-get install libboost-system-dev libboost-test-dev libboost-thread-dev libqwt-qt5-dev qtbase5-dev
git clone https://github.com/srsLTE/srsGUI.git
pushd srsGUI
mkdir build
cd build
cmake ..
make
sudo make install
popd
Install SRSLTE (SRSenb & SRSue)
pushd srsLTEmkdir build cd build cmake ../ make make test sudo make install sudo ldconfig sudo ./srslte_install_configs.sh service popd
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.
So we’ve got a functional network, but let’s dive deeper into what we can do to connect it with other networks and how things work in “the real world”.
Media Handling – OsmoMGW
The Audio/Voice (media stream) data on a call between subscribers does not go directly between the subscribers and instead needs to be proxed relayed. The reason for this is because there’s no direct link from one BTS to another BTS (even if they are served by the same BSC) and as our subscribers can move from cell to cell while on a call – which may mean moving from one BSC to another depending on where they’re heading – we need to have a single point for the audio to remain.
To handle this a Media Gateway is used, a single point for call audio to be “anchored” – meaning regardless of which BTS/BSC is serving the subscribers on either end of the call, the media will be sent by both parties to a single destination (The Media Gateway), and that destination will send the audio to the other party.
The Media gateway relays / proxies the Media Stream – the RTP packets containing the call audio. OsmoMSC provides the SDP payload containing the codecs and RTP details for the session via MGCP (Media Gateway Control Protocol) to the OsmoMGW which relays the audio.
In terms of running osmo-mgw we installed it earlier,
The only parameter you really need to change is the rtp bind-ip,
On the MGW you can also limit and restrict the codecs supported and also allow or disallow transcoding.
MNCC-SAP & Call Routing
In it’s default mode, the OsmoMSC will look at the destination a call is being sent to, and if the destination is a subscriber on the network (in it’s VLR), will route the call to that subscriber
This on-net only mode is great but it puts our network on an island – cut off from the outside world.
Calls between MSCs, to the PSTN and users everywhere else are not possible in this scenario.
3GPP defined “MNCC-SAP” (Mobile Network Call Control – Service Access Point) a protocol for handling calls to/from destinations outside of the local MSC.
When in MNCC-SAP mode all calls (even on-net calls between subscribers on the same MSC) are routed to the external MNCC-SAP, and left up to it to determine how to route the call.
Configuring Osmo-MSC to talk MNCC
As we just covered by default Osmo-MSC only switches calls internally between subscribers, so we’ll need to turn off this behaviour and isntead reconfigure it to talk MNCC-SAP.
To do this we’ll telnet / VTY into Osmo-MSC;
root@gsm-bts:/etc/osmocom# telnet localhost 4254
Welcome to the OsmoMSC VTY interface
OsmoMSC - Osmocom Circuit-Switched Core Network implementation
OsmoMSC> enable
OsmoMSC# configure terminal
OsmoMSC(config)# msc
OsmoMSC(config-msc)# mncc external /tmp/msc_mncc
OsmoMSC(config-msc)# end
OsmoMSC# cop run st
Configuration saved to /etc/osmocom/osmo-msc.cfg
After making this change we have to restart OsmoMSC;
systemctl restart osmo-msc
Now OsmoMSC will not switch calls locally, but instead when a mobile originated call comes to the MSC, it will signal to the external MNCC via the file sock at /tmp/msc_mncc,
MNCC-SAP sounds great but platform X only speaks SIP
Enter the Osmo-SIP-Connector, which takes the MNCC-SAP messages and converts them to SIP.
From here you can tie the call control functions of the MNC into any SIP software such as Kamailio, FreeSwitch, Asterisk, etc, to handle call routing, number translation, application services like voicemail and conferencing, etc, etc.
On my to-do list is to make a call between one subscriber on GSM and one on VoLTE, I’ll cover that in a subsequent post.
So anywho, let’s get Osmo-SIP-Connector setup, I’m running it on the same box as the MSC on 10.0.1.201, My softphone client is running on 10.0.1.252
root@gsm-bts:/etc/osmocom# apt-get install osmo-sip-connector
root@gsm-bts:/etc/osmocom# telnet localhost 4256
Welcome to the OsmoSIPcon VTY interface
OsmoSIPcon> enable
OsmoSIPcon# configure t
OsmoSIPcon(config)# mncc
OsmoSIPcon(config-mncc)# socket-path /tmp/msc_mncc
OsmoSIPcon(config-mncc)# exit
OsmoSIPcon(config)# sip
OsmoSIPcon(config-sip)# local 10.0.1.201 5060
OsmoSIPcon(config-sip)# remote 10.0.1.252 5060
OsmoSIPcon(config-sip)# end
OsmoSIPcon# cop run st
Configuration saved to /etc/osmocom/osmo-sip-connector.cfg
Now any Mobile Originated calls will result in a SIP INVITE being sent to 10.0.1.252 port 5060 (using UDP).
Any SIP INVITES where the request URI is a valid MSISDN @ 10.0.1.201 from 10.0.1.252 will be routed to the correct subscriber for that MSISDN.
A small note – The GSM codec is (unsurprisingly) used as the codec for GSM calls by default.
Some handsets support different codecs, but many off-the-shelf IP phones don’t include GSM support, so you may find you’re required to transcode between codecs if there is no support for the other codecs.
So now we’re able to define our call routing logic in something that speaks SIP and connect calls between multiple MSCs, VoLTE / IMS networks and fixed networks, all based on what we do with the SIP.
Local Call, Local Switch
If two subscribers on the same BSC call each other, the RTP / call audio will route to the MGW where it’s anchored.
This makes sense from a mobility standpoint, but adds load to the MGW and relies on a quality A interface connection, which may be an issue depending on what backhaul options you have.
Local Call, Local Switch is a 3GPP spec to allow the RTP / call audio to act as the RTP proxy instead of the MGW.
So now we’ve covered the basics of what’s involved let’s get some traffic on our network.
For starters we’ll need to start each of our network elements and bring up whichever BTS hardware we’re using.
In order for our calls to have audio, we’ll need to set a parameter on the Media Gateway. We’ll cover the Media Gateway in more detail down the line, but there’s one value in the MGW we’ll need to set in order to have calls working, and that’s the rtp bind-ip value. You can either set it in the config file or via VTY/Telnet on port 4243.
We’ve talked about using systemctl to start all the services, but there’s a script in the /etc/osmocom directory called osmocom-all.sh which starts all the network elements for us.
Once you’ve got all the services started I’d suggest hopping onto the OsmoBSC and enabling all the logging you can, then connecting / starting your BTS.
You should see the Abis over IP connection & OML link come up as the BTS connects to the BSC.
And then, hold your breath, power up a phone and search for networks.
All going well you’ll see OsmoMSC in the network search, select it and you should see log data flying by as the phone (“terminal”) connects to the network.
Assuming you configured the IMSI of the SIM on the HLR you should be connected to the network and showing bars on the phone.
You can check your phone number (MSISDN) by dialling the USSD code *#100#
But it’s not a network with just one phone connected, connect a second phone, check it’s phone number the same way and call from one to the other.
SMS should also just work.
And there you have it, a functional GSM network!
But this isn’t the end for us, it’s really just the beginning.
There’s still so much more to learn and work on – Over the next few weeks / months we’ll add packet data to the network with GPRS or EDGE, connect into external call routing and SMS routing interfaces, use Circuit Switched Fallback to provide voice service to users on LTE networks and roam between them.
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