== Signaling Networks: SS7 and SIGTRAN

Classic digital telephony networks (whether wired or wireless) use the
ITU-T SS7 (Signaling System 7) to exchange signaling information
between network elements.

Most of the ETSI/3GPP interfaces in the GSM and UMTS network are also
based on top of [parts of] SS7.  This includes, among others, the
following interfaces:

* _A_ interface between BSC and MSC
* _IuCS_ interface between RNC (or HNB-GW) and MSC
* _IuPS_ interface between RNC (or HNB-GW) and SGSN

NOTE: This does not include the A-bis interface between BTS and BSC.
While Abis traditionally is spoken over the same physical TDM circuits
as SS7, the protocol stack from L2 upwards is quite different (Abis
uses LAPD, while SS7 uses MTP)!

=== Physical Layer

The traditional physical layer of SS7 is based on TDM (time division
multiplex) links of the PDH/SDH family, as they were common in ISDN
networks.  Some people may know their smallest incarnation as
so-called E1/T1 links.  It can run either on individual 64kBps
timeslots of such a link, or on entire 2Mbps/1.5MBps E1/T1 links.

There are also specifications for SS7 over ATM, though it is unclear
to the author if this is actually still used anywhere.

On top of the Physical Layer is the Message Transfer Part (MTP).

=== Message Transfer Part (MTP)

MTP is the lower layer of the SS7 protocol stack.  It is comprised of
two sub-layers, called MTP2 and MTP3.

Nodes in a MTP network are addressed by their unique PC (Point Code).

A _MTP Routing Label_ is in the MTP header and indicates the
_Originationg Point Code_ (OPC) as well as the _Destination Point
Code_ (DPC) and the _Service Indicator Octet_ (SIO).  The SIO is used
to de-multiplex between different upper-layer protocol such as ISUP,
TUP or SCCP.

Routing is performed by means of routers with routing tables, similar
to routing is performed in IP networks.  Even the concept of a _point
code mask_ analogous to the _netmask_ exists.

Routers are connected with one another over one or more _Link Sets_,
each comprised of one or multiple _Links_.  Multiple Links in a
Linkset exist both for load sharing as well as for fail over purposes.

==== Point Codes

The length of point codes depends on the particular MTP dialect that
is used.  In the 1980s, when international telephony signaling
networks were established, most countries had their own national
dialects with certain specifics.

Today, mostly the ITU and ANSI variants survive.  The ITU variant uses
14bit point codes, while the ANSI variant uses 24 bit point code
length.

Point Codes can be represented either as unsigned integers, or
grouped.  Unfortunately there is no standard as to their
representation.  In ITU networks, the _3.8.3_ notation is commonly
used, i.e. one decimal for the first 3 bits, followed by one decimal
for the center 8 bits, followed by another decimal for the final 3
bits.

Example:: The Point Code *1.5.3* (in 3.8.3 notation) is 1*2^11^ + 5*2^3^ + 3 = *2091 decimal*.

=== Higher-Layer Protocols

There are various higher-layer protocols used on top of MTP3, such as
TUP, ISUP, BICC as well as SCCP.   Those protocols exist side-by-side
on top of MTP3, similar to e.g. ICMP, TCP and UDP existing
side-by-side on top of IP.

In the context of cellular networks, SCCP is the most relevant part.

=== Signaling Connection Control Part (SCCP)

SCCP runs on top of MTP3 and creates something like an overlay network
on top of it.  SCCP communication can e.g. span multiple different
isolated MTP networks, each with their own MTP dialect and addressing.

SCCP provides both connectionless (datagram) and connection-oriented
services.  Both are used in the context of cellular networks.

==== SCCP Addresses

SCCP Addresses are quite complex.  This is due to the fact that it is
not simply one address format, but in fact a choice of one or multiple
different types of addresses.

SCCP Addresses exist as _Calling Party_ and _Called Party_ addresses.
In the context of connectionless datagram services, the sender is
always the Calling Party, and the receiver the Called Party.  In
connection-oriented SCCP, they resemble the initiator and recipient of
the connection.

.SCCP Address Parts
[options="header",cols="10%,20%,70%"]
|====
|Acronym|Name|Description
|SSN|Sub-System Number|Describes a given application such as e.g. a
                       GSM MSC, BSC or HLR.  Can be compared to port
                       numbers on the Internet
|PC|Point Code        |The Point Code of the underlying MTP network
|GT|Global Title      |What most people would call a "phone number".
                       However, Global Titles come in many different
                       numbering plans, and only one of them (E.164)
                       resembles actual phone numbers.
|RI|Routing Indicator |Determines if message shall be routed on PC+SSN
                       or on GT basis
|====

==== Global Titles

A Global Title is a (typically) globally unique address in the global
telephony network.  The body of the Global Title consists of a series
of BCD-encoded digits similar to what everyone knows as phone numbers.

A GT is however not only the digits of the "phone number", but also
some other equally important information, such as the _Numbering Plan_
as well as the _Nature of Address Indication_.

.Global Title Parts
[options="header",cols="10%,20%,70%"]
|====
|Acronym|Name|Description
|GTI|Global Title Indicator|Determines the GT Format. Ranges from no
                            GT (0) to GT+TT+NP+ES+NAI (4)
|NAI|Nature of Address Indicator|Exists in GTI=1 and is sort of a mixture of TON + NPI
|TT|Translation Type      |Used as a look-up key in Global Title Translation Tables
|NP|Numbering Plan        |Indicates ITU Numbering Plan, such as E.164, E.212, E.214
|ES|Encoding Scheme       |Just a peculiar way to indicate the length of the digits
|- |Signals               |The actual "phone number digits"
|====

For more information about SCCP Addresses and Global Titles, please
refer to <<itu-t-q713>>


==== Global Title Translation (GTT)

Global Title Translation is a process of re-writing the Global Title
on-the-fly while a signaling message passes a STP.

Basically, a SCCP message is first transported by MTP3 on the MTP
level to the Destination Point Code indicated in the MTP Routing
Label.  This process uses MTP routing and is transparent to SCCP.

Once the SCCP message arrives at the MTP End-Node identified by the
Destination Point Code, the message is handed up to the local SCCP
stack, which then may implement Global Title Translation.

The input to the GTT process is

* the destination address of the SCCP message
* a local list/database of Global Title Translation Rules

The successful output of he GTT includes

* A new Routing Indicator
* The Destination Point Code to which the message is forwarded on MTP
  level
* a Sub-system Number (if RI is set to "Route on SSN")
* a new Global Title (if RI is set to "Route on GT"), e.g. with translated digits.

Between sender and recipient of a signaling message, there can be many
instances of Global Title Translation (up to 15 as per the hop
counter).

For more information on Global Title Translation, please refer to
<<itu-t-q714>>.


==== Peculiarities of Connection Oriented SCCP

Interestingly, Connection-Oriented SCCP messages carry SCCP Addresses
*only during connection establishment*.  All data messages during
an ongoing connection do not contain a Called or Calling Party
Address.  Instead, they are routed only by the MTP label, which is
constructed from point code information saved at the time the
connection is established.

This means that connection-oriented SCCP can not be routed across MTP
network boundaries the same way as connectionless SCCP messages.
Instead, an STP would have to perform _connection coupling_, which is
basically the equivalent of an application-level proxy between two
SCCP connections, each over one of the two MTP networks.

This is probably mostly of theoretical relevance, as
connection-oriented SCCP is primarily used between RAN and CN of
cellular network inside one operator, i.e. not across multiple MTP
networks.

=== SIGTRAN - SS7 over IP Networks

At some point, IP based networks became more dominant than classic
ISDN networks, and 3GPP as well as IETF were working out methods in
which telecom signaling traffic can be adapted over IP based
networks.

Initially, only the edge of the network (i.e. the applications talking
to the network, such as HLR or MSC) were attached to the existing old
SS7 backbone by means as SUA and M3UA.  Over time, even the links of
the actual network backbone networks became more and more IP based.

In order to replace existing TDM-based SS7 links/linksets with SIGTRAN,
the M2UA or M2PA variants are used as a kind of drop-in replacement
for physical links.

All SIGTRAN share that while they use IP, they don't use TCP or UDP
but operate over a (then) newly-introduced Layer 4 transport protocol
on top of IP: SCTP (Stream Control Transmission Protocol).

Despite first being specified in October 2000 as IETF RFC 2960, it
took a long time until solid implementations of SCTP ended up in
general-purpose operating systems.  SCTP is not used much outside the
context of SIGTRAN, which means implementations often suffer from bugs,
and many parts of the public Internet do not carry SCTP traffic due to
restrictive firewalls and/or ignorant network administrators.

==== SIGTRAN Concepts / Terminology

Like every protocol or technology, SIGTRAN brings with it its own
terminology and concepts.  This section tries to briefly introduce
them.  For more information, please see the related IETF RFCs.

===== Signaling Gateway (SG)

The Signaling Gateway (SG) interconnects the SS7 network with external
applications.  It translates (parts of) the SS7 protocol stack into an
IP based SIGTRAN protocol stack.  Which parts at which level of the
protocol stack are translated to what depends on the specific SIGTRAN
dialect.

A SG is traditionally attached to the TDM-Based SS7 network and offers
SIGTRAN/IP based applications a way to remotely attach to the SS7
network.

A SG typically has STP functionality built-in, but it is not
mandatory.

===== Application Server (AS)

An Application Server is basically a logical entity representing one
particular external application (from the SS7 point of view) which is
interfaced with the SS7 network by means of one of the SIGTRAN
protocols.

An Application Server can have one or more Application Server Processes
associated with it.  This functionality can be used for load-balancing or
fail-over scenarios.

===== Application Server Process (ASP)

An Application Server Process represents one particular SCTP
connection used for SIGTRAN signaling between an external application
(e.g. a BSC) and the Signaling Gateway (SG).

One Application Server Process can route traffic for multiple
Application Servers.  In order to differentiate traffic for different
Application Servers, the Routing Context header is used.

==== SIGTRAN variants / stackings

SIGTRAN is the name of an IETF working group, which has released an
entire group of different protocol specifications.  So rather than one
way of transporting classic telecom signaling over IP, there are now
half a dozen different ones, and all can claim to be an official IETF
standard.

FIXME: Overview picture comparing the different stackings

===== MTP3 User Adaptation (M3UA)

M3UA basically "chops off" everything up to and including the MTP3
protocol layer of the SS7 protocol stack and replaces it with a stack
comprised of M3UA over SCTP over IP.

M3UA is specified in <<ietf-rfc4666>>.

M3UA is the SIGTRAN variant chosen by 3GPP for A, IuCs and IuPS
interfaces over IP.

===== SCCP User Adaptation (SUA)

SUA basically "chops off" everything up to and including the SCCP
protocol layer of the SS7 protocol stack and replaces it with a stack
comprised of SUA over SCTP over IP.

This means that SUA can only be used for SCCP based signaling, but not
for other SS7 protocols like e.g. TUP and ISUP.

SUA is specified in <<ietf-rfc3868>>.

===== MTP2 User Adaptation (M2UA)

M2UA is specified in <<ietf-rfc3331>>.

NOTE: M2UA is not supported in Osmocom SIGTRAN up to this point.  Let
us know if we can implement it for you!

===== MTP2-User Peer-to-Peer Adaptation (M2PA)

M2PA is specified in <<ietf-rfc4165>>.

NOTE: M2PA is not supported in Osmocom SIGTRAN up to this point.  Let
us know if we can implement it for you!


==== SIGTRAN security

There simply is none.  There are some hints that TLS shall be used
over SCTP in order to provide authenticity and/or confidentiality for
SIGTRAN, but this is not widely used.

As telecom signaling is not generally carried over public networks,
private networks/links by means of MPLS, VLANs or VPNs such as IPsec
are often used to isolate and/or secure SIGTRAN.

Under no circumstances should you use unsecured SIGTRAN with
production data over the public internet!

==== IPv6 support

SCTP (and thus all the higher layer protocols of the various SIGTRAN
stackings) operates on top of both IPv4 and IPv6.  As the entire
underlying IP transport is transparent to the SS7/SCCP applications,
there is no restriction on whether to use SIGTRAN over IPv4 or IPv6.

==== SCTP multi-homing in SIGTRAN

SCTP, unlike more traditional IP L4 protocols (TCP, UDP) doesn't work
based on a _connection_ between source IP:port and Destination IP:port.

Instead, SCTP creates _associations_ between two endpoints, both of which
can have any number of IP addresses.  This means that in case of
network outage, traffic can continue to flow through any of the IP
addresses of that association.

The Linux kernel by default advertises all IP addresses of the local
system to the peer.  This can be seen when inspecting the SCTP INIT
chunk e.g. in wireshark.  While this may be a reasonable default in some
use cases, it is not always the best idea.  Imagine addresses of
internal/private IP networks, for example local bridge devices between
lxc or docker containers, or local VMs.  Such addresses have no
significance beyond the local machine.

Subsequently, libosmo-sigtran allows the user to explicitly select which
local IP addresses shall be used in SCTP multi-homing for the SIGTRAN
associations it manages.  The user can achieve this by specifying
multiple `local-ip` VTY commands within one `asp` (SCTP client role) or
within one `listen m3ua 2905` (SCTP server role).

==== SCTP Primary Address

SCTP has the concept of "primary address" in an association. The primary address
is a remote address selected from those announced by the peer, and it is the
"active" one chosen to transmit user data. The other remote addresses, that are
not used, are kept as backups. They are in general only used to transmit user
data whenever the SCTP implementation decides to change the primary address, be
it due to user policy configuration change or due to the previous primary link
becoming unusable. Only confirmed remote addresses (through HEARTBEAT mechanism)
are electable to be used as primary address.

By default, the Linux kernel SCTP stack implementation will probably take the
first remote address provided at connect() time in order to start the initial
handshake, and continue with next provided remote addresses if the first one
fails to confirm the handshake. The remote address which successfully confirmed
the handshake is then used as a primary address (since it's likely the only
confirmed so far), and will be kept until the link is considered down.

Some deployment setups may have requirements on preferred links to be used when
transmitting data (eg. network setups with primary and secondary paths). This
can be accomplished by explicitly notifying the kernel to use one of the remote
addresses through the SCTP_PRIMARY_ADDR sockopt, plus monitoring the address
availability changes on the socket and re-enforcing the primary address when it
becomes available again. This is supported in the Osmocom SIGTRAN stack by using
the `primary` parameter in one of the `remote-ip` commands under the `asp` node:

----
cs7 instance 0
 asp my-asp 2905 0 m3ua
  remote-ip 10.11.12.13
  remote-ip 16.17.18.19 primary <1>
  ...
----
<1> Use 16.17.18.19 as primary address for the SCTP association. User data will
be in general transmitted over this path.

==== SCTP Peer Primary Address

The SCTP extension ASCONF (RFC5061) allows, when negotiated and supported by
both peers, to dynamically announce to the peer the addition or deletion of IP
addresses to the association. It also allows one peer announcing to the other
peer the desired IP address it should be using as a primary address when sending
data to it.

In the Linux kernel SCTP stack, this is accomplished by setting the sockopt
SCTP_SET_PEER_PRIMARY_ADDR, which will trigger an ASCONF SCTP message to the
peer with the provided local IP address. This is supported in the Osmocom
SIGTRAN stack by using the `primary` parameter in one of the `local-ip` commands
under the `asp` node:

----
cs7 instance 0
 asp my-asp 2905 0 m3ua
  local-ip 10.11.12.13
  local-ip 16.17.18.19 primary <1>
  ...
----
<1> Announce 16.17.18.19 to the peer as the primary address to be used when
transmitting user data to us.

In order to be able to use this feature, the SCTP association peer must support
the ASCONF extension. The extension support is negotiation during the INIT
handshake of the association. Furthermore, for ASCONF features to work properly,
the assoc also needs to announce/use the AUTH extension, as per RFC5061 section
4.2.7. Otherwise, the peer receiving an SCTP INIT with
`ExtensionFeatures=ASCONF,ASCONF_ACK`` but without AUTH, will reject the
association with an ABORT since it's not complying with specifications (this
behavior can be tweaked through sysctl "net.sctp.addip_noauth_enable").

As of the time of writing this documentation (linux 6.4.12) and since basically
ever, those extensions are runtime-disabled by default. They can be enabled per
socket using the kernel sockopts SCTP_ASCONF_SUPPORTED and SCTP_AUTH_SUPPORTED,
and that's what the Osmocom stack is currently doing for all SCTP sockets.
However, those sockopts are farily new (linux v5.4), which means user running
older kernels will see in the logs setting those sockopts fail, but connection
will keep ongoing, simply without those features available (so setting `primary`
in the configuration won't have any effect here).
On those older kernels, if this feature is still desired, it can be used
by means of enabling the SCTP extensions in all socket system-wide through sysctl:
----
net.sctp.auth_enable=1
net.sctp.addip_enable=1
----

==== SCTP INIT Parameters

Several SCTP INIT parameters can be configured through VTY, which will be passed
to the Linux Kernel SCTP stack and used whenever an association is being
established.

On the client side (see <<sctp_role>>), the parameters are configured in the `asp` node:
----
cs7 instance 0
 asp my-asp 2905 0 m3ua
  sctp-role client
  sctp-param init num-ostreams 250 <1>
  sctp-param init max-instreams 300 <2>
  sctp-param init max-attempts 3 <3>
  sctp-param init timeout 10000 <4>
  ...
----
<1> The number of streams from the server to the client. This value is
transmitted during SCTP INIT ACK packet.
<2> Announce to the server that a maximum of up to 300 inbound SCTP streams are
supported. This value is transmitted during SCTP INIT packet.
<3> Initial SCTP handshake will be attempted 3 times before considering the
connection failed.
<4> Retransmit an SCTP INIT message after 10000 ms if no answer is received.

On the server side (see <<sctp_role>>), the parameters are configured in the `listen` node:
----
cs7 instance 0
 asp my-asp 2905 0 m3ua
  sctp-role server
 listen m3ua 2905
  sctp-param init num-ostreams 250 <1>
  sctp-param init max-instreams 300 <2>
  ...
----
<1> Announce to the server that up to 250 outbound SCTP streams (server to
client) may be requested. This value is transmitted during SCTP INIT packet, and
should be equal or lower to the `max-instreams` value received from the client
during SCTP INIT packet.
<2> Announce to the server that a maximum of up to 300 inbound SCTP streams are
supported. This value is transmitted during SCTP INIT ACK packet.

[[sctp_role]]
==== SCTP role

The _SCTP role_ defines which of the two L4 protocol roles SCTP assumes:

* The _SCTP server_ role binds to a local port and handles incoming
  connections from clients
* The _SCTP client_ role connects to a remote SCTP sever.

==== M3UA/SUA role

The _M3UA role_ (or _SUA role_) determines which role a given peer of a
M3UA connection implements.  3GPP specifies the following role:

* _SGP_ (Signaling Gateway): The entity connected to the larger SS7
  network
* _ASP_ (Application Server Process): A client application that connects
  to the SGW to talk to the SS7 network
* _IPSP_ (IP Server Process): M3UA in point-to-point mode

Osmocom (libosmo-sigtran) implements both the SGP and ASP roles, but not
the IPSP role.

==== Traffic Modes in SIGTRAN

Whenever an AS consists of multiple ASPs, the traffic mode expresses how
messages are distributed between those ASPs.

* _Override_: There is always one active ASP and multiple hot standby
  ASPs.  If the active ASP fails, one of the remaining ASPs will become
  the new active ASP.
* _Loadshare_: The messages will be distributed between the different
  ASPs in a way to distribute the load among them.  Details are
  implementation specific.
* _Broadcast_: A copy of every incoming signaling message is sent to
  _all_ the ASPs in broadcast traffic mode.

Osmocom (libosmo-sigtran) implements all above-mentioned traffic modes.
