S38.3115 Signaling Protocols – Lecture Notes

S38.3115 Signaling Protocols – Lecture Notes

Lecture 1: Introduction to Signaling

Purpose of signaling....................................................................................... 1

Call state......................................................................................................... 3

Signaling and mobility................................................................................... 4

Digital transmission and structure of PCM system........................................ 4

Frame synchronization in PCM system ..................................................... 5

Allocation of functions in different networks................................................ 7

Voice coding .............................................................................................. 7

Switching in circuit networks .................................................................... 8

Signaling and routing................................................................................. 9

Types of exchanges.................................................................................. 10

A structure of an exchange for circuit networks...................................... 10

Fault-tolerance requirements on switching systems ................................ 12

What happens to switch function if calls are carried over an IP network 13

Generalization: telephony paradigm ............................................................ 13

Differences between public and private switched networks........................ 14

Internet Architecture for Voice over IP ....................................................... 15

Network intelligence vs. terminal intelligence ............................................ 16

Scope of our discussion on this course ........................................................ 17

Prerequisites for the course

We assume knowledge of software specification, design and implementation and some hands-on experience on IP or circuit switching equipment.

Hopefully, at least part of the software experience should be from a real time environment. The hands-on experience can be gained on S38 laboratory courses. In the past those with little or no software experience have had great difficulties in comprehending what we will be talking about. I hope these lecture notes will make things clearer and at least a bit easier to learn.

Purpose of signaling

The purpose of signaling is to set-up, supervise and tear down calls.

Traditional networks such as the PSTN (Public Service Telephone Network

(yleinen puhelinverkko)) support the setting up of voice calls where an audio channel is established between two users or subscribers. Modern networks may support the setting up of multimedia calls that in addition to audio or voice transfer other types of media, like video or data. lecture 1

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S38.3115 Signaling Protocols – Lecture Notes

Let us first define a few key terms: circuit switching, packet switching,

connection oriented and connectionless. First, circuit switched networks are always connection oriented. In such networks, the units of data that are switched in network nodes do not carry address information and a switching fabric state is established in each node between an incoming circuit and an outgoing circuit such that data units will continuously flow from the incoming to the outgoing circuit. In these networks only having first established a connection from end to end, users of the network can transfer data. Usually, we establish a bidirectional connection. In PSTN, these connections have a constant capacity of 64 kbit/s and the units of data that are switched are bytes.

Packet switched networks may be either connection oriented or connection less. Packets always carry a header including some address information plus a payload. In a connection oriented packet network, the packets will carry address info that has local significance. As a result, an end-to-end virtual

connection needs to be established before two users can communicate. This is the case for example in ATM.

In a connectionless packet switched network, packets carry an address or addresses that have global significance, i.e. are globally unique. An example is the Internet based on the IP protocol. For end to end connectivity, a sender just needs to know a globally unique address of the receiver and data can start flowing from one user to another. As we will see, in practice the Internet has a lot of limitations for sending packets between any two Internet users and besides globally unique addresses makes use of private IP addresses as well..

The term “call” refers to a phenomenon that involves subscriber A - calling party or a caller (different terms are used for historical reasons), the caller’s device, a chain of transmission systems and network nodes, subscriber B

(called party or callee) and the callee’s device. A term that is nowadays also used instead of “call” is session. For setting up the call signaling is needed and within a call some communication between the parties takes place.

Let us now define what is signaling. Signaling is the transfer of control

information for the purpose of setting up, supervising and tearing down calls

between users in a network.

From the definition one can see that we are dealing with a control system where the object of control is the resources in the network and the control intelligence is called “call control” residing in several network nodes. The language that is used to carry control information between different call control elements is called signaling.

Network technologies used to carry calls differ. We may have a circuit switched, connection oriented network (e.g. PSTN or ISDN or GSM) or a packet switched connectionless network (Internet or an IP network). lecture 1

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S38.3115 Signaling Protocols – Lecture Notes

Physically, the resources that are used to build the networks also differ greatly.

Signaling is different in different networks.

What is common to all known networks in terms of signaling is that we need

to locate the called party, figure out if he or she wishes to engage in a call and

to agree on a minimum set of technical parameters for the session. These functions are needed independent of technology. To create the agreement between a caller and the callee, the parties need to agree on things like: audio coding method and speed, video coding method and speed etc.

Characteristic of a call is that resources such as circuits (e.g. time-slots in the

PCM transmission systems) are reserved in the network for the duration of the call. E.g. in PSTN, ISDN and GSM networks a voice circuit is established end to end. The end to end connection has several legs that are clued together

(switched) at exchanges on the call path. It may be that for a period during a call such as part of the establishment phase some special resources are needed.

Examples are tone generators that can send for example dial tone or busy tone to the caller. Older signaling systems used to require signal receivers and signal senders. From a network point of view it is important that exchanges keep accurately track of the use of all these resources, so they are reserved for the call and released when they are no longer needed. From the need to connect e.g. tone generators to the voice circuit it follows that establishment of a circuit connection takes many switching actions and that call control in circuit networks takes a lot of micro-management of network resources during a call.

To further characterize signaling, we note that from a network node point of view, relating to a single call it must exchange control information on the incoming side (towards the caller) and on the outgoing side (towards the callee), each neighbor works in its own pace while the node needs to make decisions based on its own data, or data it may need from other control nodes and keep track of the resources reserved for the call.

Call state

The result is that a call has state that is created at the beginning of the call and changes on events during the call. The state information on a call is needed in the callers device, in each exchange or signaling node on the signaling path and in the callee’s device. Moreover, it is convenient in a network signaling node to separate the following concerns: incoming signaling, incoming call control, outgoing call control and outgoing signaling.

Each of these functions has its own state. Incoming signaling state keeps track of the sequence of signals on the incoming side, incoming call control is responsible for making a routeing decision: i.e. based on called party number it will decide in which direction the call should proceed from the current node, lecture 1

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S38.3115 Signaling Protocols – Lecture Notes outgoing call control is responsible for outgoing side resources and outgoing signaling state reflects the state of communication with the next hop neighbor on the call path.

In an IP network the resources that may need to be controlled by signaling differ from Circuit networks such as PSTN. Related to signaling we note that, if the Internet were as it was initially designed, an end-to-end transparent network for any user to communicate with any other user, the job of signaling would be simple. For such ideal network, all that would be needed is to locate the callee, establish a wish to communicate and agree on codecs and other technical parameters for communication between the user devices.

However, the current Internet is not an ideal end-to-end transparent network.

Actually, one might say that the Internet does not have B-subscribers because users are behind firewalls, use private IP-addresses etc. So, a call over an IP network may require opening temporary holes in firewalls and finding means to traverse Network Address Translators that sit on address space boundaries.

Signaling and mobility

It is convenient for users to be reachable where-ever they are. For this purpose the network needs to keep track of user’s locations. For example, in GSM the mobile makes location updates regularly automatically. The network has a

Home Location Register, a database that stores the location information with the accuracy of a Visitor Location Register that is inside a Mobile Switching

Center taking care of the users roaming in a certain geographical area.

Also the transfer of location and other mobility related control information falls under the definition of signaling although the transfer may take place while there is no call.

Digital transmission and structure of PCM system

In order to understand why call state is needed and to understand the architectural constraints that needed to be taken into account when digital signaling systems were designed, we must understand the transmission plant over which calls and signaling are carried.

Pulse code modulation or PCM was the first digital transmission system. It was invented in the 1930’s but implementation became feasible only late

1960’s. For example Nokia used to be in the business of manufacturing telephone cables and during 1960’s it decided to add value to their cables by designing a digital transmission system based on PCM. This is the origin of

Nokia’s path to electronics and telecom equipment manufacturing. Nokia’s first PCM system went to live use in a network in 1968. lecture 1

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S38.3115 Signaling Protocols – Lecture Notes

During 1980’s a move to the SDH (synchronous digital hierarchy) started.

Inside SDH frame PCM signals are still carried. Also, a SDH signal is usually broken into PCM signals before it is interfaced to an exchange.

The starting point of the PCM system is that voice is sampled 8000 times per second in order to transfer the narrow band of less than 3.4 kHz. Recall the

Nyqvist theorem of digital transmission: in order to transfer and recover a signal with a maximum frequency of N, the analogue signal must be sampled at double that rate. In the PCM system each voice sample is carried with 8 bits.

This gives the basic speed of one channel that is used in the PCM system:

8000 × 8 = 64 kbit/s. A channel in PCM can be viewed as a continuous digital signal at the speed of 64 kbit/s.

PCM is a time division multiplexing system (TDM). This means that on a single cable, the system multiplexes many channels that can each carry for example one voice signal of 64 kbit/s. In ETSI networks, the first order multiplexed PCM signal has 32 channels of which one is used for frame synchronization and the rest in principle can be used for voice or data. In practice, from the era of analogue signaling usually one of the 31 channels is allocated for signaling, so 30 channels remain for voice or user’s data. The basic multiplexed 32 channel PCM signal is called E1. The transmission speed is 32 × 64kbit/s = 2048 kbit/s. Often we just talk about a 2Mbit/s signal.

For understanding how PCM works we must look at the signal from the receiver’s perspective. Sending bits is easy. The hard part is receiving and making sense of the received signal.

The E1 signal is a series of bits. Bits are represented in a line code called

HDB3, we will talk about it later on the ISDN lecture. For the time being, let us assume that there is a way of receiving bits that can be efficiently implemented on silicon. How can the receiver decide where are the channel boundaries in the series of bits? The answer is framing.

Frame synchronization in PCM system

The PCM signal can be broken down into timeslots. A timeslot is a placeholder for a voice sample or user’s data of 8 bits or for some other 8 bits.

In the PCM system a sequence of one synchronization timeslot of 8 bits plus the timeslots for the following 31 channels is called a frame. The length of the frame is 32 × 8bits = 256 bits. A PCM frame is presented in Figure 1.1.

0 1 2 3

31 0 1 2 3 4 5 time,


0 62,5 125

Figure 1.1: A PCM signal. lecture 1

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S38.3115 Signaling Protocols – Lecture Notes

In Figure 1.1, timeslots are numbered from 0 to 31 and the figure shows how after the first frame the next follows immediately after. The Figure also shows timing of the signal. The length of the PCM frame in time is 1/8000 s = 125


The time duration of single timeslot is 125/32 = 3,9

µs. The length of a bit in time is 1/2,048 = 3,9/8 = 0,488


So, how does the receiver figure out where the frame and channel boundaries are? It is agreed that the contents of timeslot zero are used for frame synchronization and that the contents in each other timeslot zero are c0011011.

The frame in which this appears is called an even frame (frame number 0, 2,

4…). Having received this bit combination the receiver counts to 63 and looks if the next 8 bits are again the same c0011011 (we call this a frame mark). If yes, the received keeps doing counting and receiving and if it sees the same bit combination several times at this distance from each other, it decides that it has found the frame boundaries and consequently just by counting it is also able to spot the timeslot boundaries.

What if one of the users would also send the c0011011 for a long time? While operators were carrying only digitally encoded analogue signals of voice or data encoded with analogue modems this would not have been possible.

However, when the Integrated Services Digital Network was introduced, it meant that users are offered a transparent 64 kbit/s channel. Channel transparency means that a user can send any bit combination for any length of time. Due to ISDN it would have been possible for a user, for the sake of prank if nothing else to keep sending the frame mark and if a receiver looses its frame sync and needs to recover, it could accidentally have synchronized to the user’s mark instead of synchronizing to the timeslot zero. This was unacceptable and after close to 20 years of using the PCM system it had to be upgraded for ISDN networks. The upgrade was that the first bit of the frame

mark was redefined to carry a checksum of 8 bits calculated over the previous

8 frames. The checksum bit is shown by “c” in our notation for of the frame mark. The result is that besides the base frame, a PCM signal in ISDN also has a multi-frame of 16 frames.

The length of the multi-frame is 16 × 256 bits = 4096 bits.

In time the length is 16 × 125

µs = 2ms.

The result is that resynchronization of a PCM receiver to an ISDN compatible

PCM signal may take several milliseconds.

In PSTN with analogue signaling a PCM has another multi-frame structure that is used for Channel Associated signaling (CAS). This is based on time-slot

16 that is sub-multiplexed to carry signaling bits for voice timeslots on the same PCM line. This CAS related multi-framing is completely independent of the multi-framing for ISDN. Even if both of these multi-frame structures are used on the same PCM connection, they are independent. If multi-framing or lecture 1

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S38.3115 Signaling Protocols – Lecture Notes frame sync is lost the alarm system may produce alarms related to both at the same time and these need to be correlated but otherwise, really the two multiframings are independent.

To conclude the discussion of PCM: it carries raw uni- or bi-directional bitstreams called channels in timeslots. A channel may be allocated to a user or a telephone conversation and it carries no overhead for management of other purposes.

Allocation of functions in different networks

In this Section we discuss the allocation of different functions related to signaling and telephony as a whole in different places in networks and user devices.

Voice coding

Packet networks such as the Internet differ from Circuit networks such as

PSTN in where the responsibility for coding and decoding of voice resides. In

Packet networks the connected devices are always computers of some sort while the network just moves packets. Therefore, it is natural that voice coding and decoding responsibility lies with the user devices.

PSTN phones can be simple electromechanical devices. In PSTN, such analogue phones send voice onto the local loop as an analogue electrical signal and it is typically encoded into a digital form on the first circuit board to which the local loop is connected. This board is called a line card or a subscriber line

card. In the opposite direction decoding of voice takes place on the same card.

The fact that encoding and decoding of voice take place in the network lead to just two, one for ETSI and a slightly different variant for ANSI countries, voice coding standards for digital PSTN network: G.711 A-law and

µ-law. If each operator would use its own voice coding method, on network boundaries a lot of equipment would be needed to decode and encode between the different standards – this would obviously make no sense from an economic point of view because the cost of such equipment would have to be born by the operators.

GSM and ISDN networks were designed with the requirement of smooth interworking with PSTN that at the beginning of GSM and ISDN had more than 500 million subscribers connected while the new networks had just a few.

The design decision in ISDN was that the G.711 codec was moved from the line card in the exchange to the ISDN phone.

In GSM initially a single codec was adopted and an element called the

Transcoder was placed between the Base Station Controller and the Mobile

Switching Center to map between G.711 and GSM voice coding. By adopting lecture 1

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S38.3115 Signaling Protocols – Lecture Notes a new codec, air interface capacity was saved, because the GSM codec uses some 13 kbit/s while the PSTN, G.711 uses 64kbit/s for a single voice stream.

So, this new coding method (GSM codec) was restricted just to GSM access while the network itself still continued to use the same old G.711 inherited from digital PSTN. Later on, additional coding methods were introduced to

GSM networks, such as Enhanced Full Rate (EFR) and half rate codecs. All these are supported by the Transcoder element and mapped to G.711 for smooth interworking with the PSTN. When calls from mobile to another mobile became commonplace, the so called Transcoder Free Operation was introduced. This means that under the control of signaling, the GSM switching centers can tell the Transcoders at the Mobile Originated (MO) and Mobile

Terminated (MT) sides to ignore the mapping to G.711. The result is that the end to end circuit connection uses only 2 bits out of each 8 in each timeslot of

64 kbit/s connection. Dropping the transcoding to G.711 is useful even if one does not save core network capacity because transcoding always degrades voice quality.

Sometimes in networks elements called Media Gateways are needed. These have a similar function as the Transcoder in GSM. A Media Gateway supports several coding standards and is capable of mapping between them in real time.

An example where a media gateway is needed is on the boundary of an IP network that supports Voice over IP and a Circuit Switched Network (PSTN,

ISDN, GSM). In a Transcoder on a physical level all interfaces are based on the PCM standard for digital transmission (2Mbit/s lines) while in a Media

Gateway the physical interfaces may include PCM, SDH, ATM, Ethernet in different variants etc.

Switching in circuit networks

Exchanges in circuit networks are responsible for switching the voice circuits through. For this purpose, all voice circuits are interfaced in an exchange into the Switching Fabric through exchange terminals. The exchange terminals take care of PCM frame synchronization and monitoring of the quality of the received PCM signals. An example is a switching fabric that has 8192 PCM interfaces of 2Mbit/s each. Total switching capacity of the Fabric would be

16Gbit/s. Each PCM has 32 timeslots each 64 kbit/s in both directions of which 30 can be used to carry voice signals. Each direction has to be switched separately in the switching fabric. Each timeslot carries a bit stream on

64kbit/s that, for example, supports the transfer of 8000 voice samples of 8 bits each per second.

Also things like tone generators and devices that need to process the contents of timeslots are interfaced to the same Switching Fabric. These also include all control computers of the switching system: since these need to process signaling that is carried over PCM lines, it is convenient to attach control computers to the Fabric with internal PCM lines. Establishing a signaling link lecture 1

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S38.3115 Signaling Protocols – Lecture Notes from a control computer in an exchange to a control computer in the neighboring exchange means that a PCM timeslot between the exchanges needs to be reserved in both directions and through-connected in both exchanges to the internal PCM line that is connected to the appropriate control computer at each exchange.

It is a property of a so called non-blocking PCM switching fabric that it can map any incoming timeslot to any outgoing timeslot provided the outgoing timeslot is not in use already. Another property of the fabric is that no outgoing timeslot “hangs in the air” i.e carries nothing. An outgoing timeslot always has some signal in it and in the absence of anything else to carry, an idle tone is typically sent from the tone generator. It is also typical of modern switching fabrics that they are able to multicast any incoming timeslot into any number of outgoing timeslots.

Switching a timeslot means that the exchange writes into the control memory of the switching fabric an instruction to map a particular incoming timeslot to a particular outgoing timeslot. The switching can be unidirectional or bidirectional. The latter is needed for voice and requires two write operations into the control memory. Unidirectional switching is needed for example during a call to set up the voice path to the caller from the callee so that the exchange closest to the callee can send for example a busy tone or an announcement about the state of the callee to the caller. At this time, when the callee has not answered, the exchange closest to the caller typically wishes to keep the caller from using the voice path for any useful purpose because charging starts only when B-subscriber has answered and the answer signal has been received. Therefore, the forward call path, in the exchange closest to the caller, is not through-connected until the reception of answer signal.

Signaling and routing

In circuit networks we talk about routeing. In packet networks such as the

Internet we talk about routing. Routeing and routing are the process of

defining the path that the call, flow or packet takes through the network from

the origin to the destination. Packet routing in itself does not need signaling. In

IP networks packets are forwarded in routers based on forwarding tables that are created using dynamic routing protocols to collect information about the network topology. Once forwarding tables are in place it is possible to send packets from origin to destination. Because forwarding decisions are made packet by packet, when a forwarding entry is changed, all packet flows making use that entry will change direction.

In case of circuit switched networks signaling brings input data for routeing decisions. The data includes: the type of service requested, dialed digits etc. A routeing decision is implemented in a switching node by writing a command into the control memory of the switching fabric. User data forwarding in lecture 1

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S38.3115 Signaling Protocols – Lecture Notes switched networks is the same as switching. Routeing in circuit networks is based on static configuration information. There are no dynamic routing protocols for these networks. When route tables are changed by the operator, existing call paths are not affected. Rather the change affects only fresh calls.

Types of exchanges

We classify Exchanges based on their position on the call path. An exchange to which subscribers are physically connected to is called a local switch

(Central Office or Class 5 switch in US). A switch to which no subscribers are connected to is a transit switch. Transit switches can further be classified to

long distance and international switching centers. The circuits between exchanges are called trunks. Therefore, also the term trunk switch is sometimes used.

For a given call, the local switch or exchange closest to the caller is called the

originating exchange and the local exchange to which the callee is connected to is called the terminating exchange.

When digital switches were first introduced, it was a serious design challenge to make them big enough for the purpose of allowing building convenient network topologies. One can easily calculate that if it is possible to build switching nodes with the capacity in the area of tens of thousands of users, networks of any size can be built. When even bigger switching nodes became possible, it became even easier to build networks with tens and hundreds of millions of customers.

During the 1990’s Moore’s law took care of the problem. The result was that in digital circuit networks operators moved to larger node size. This helped to reduce software and other maintenance costs. Large switching node size led also to blurring of the categories of switching nodes. A large local switch may be used as a long distance and also as an international switch provided that the vendor is able to provide a rich enough software build for the node. A modern large switching node, in addition to supporting originating and terminating traffic to some users, may support transit traffic for users that are connected to other switching nodes.

A structure of an exchange for circuit networks

Let us return to the example of an exchange with a Switching Fabric of 8192

PCM lines. It might be connected to several tens of other exchanges. A single control computer would most likely not be enough to take care of all signaling and call control load. One way of splitting the load (used e.g. in DX 200 of

Nokia) is to allocate a fraction of those 8k PCM lines to a control computer in such a way that a prevalent or a single signaling system type is in use for calls using all PCMs of the fraction. E.g. 64, 128 or 256 PCMs all using the Number

7 signaling (we will describe this signaling system later in detail) would be lecture 1

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S38.3115 Signaling Protocols – Lecture Notes taken care of by a single control computer. Because between the control computer and the PCM lines there is the switching fabric, it is possible to remap those PCM –lines to another control computer for example if the original computer were to fail or needs to be taken out of use for the purpose of upgrading its software.

Figure 1.2 shows a high level structure of a local exchange or a switching system that provides access for subscribers. lecture 1







Local loop or radio

Control part

Figure 1.2: A high level structure of a digital switching system

The subscriber lines are connected to line cards in the subscriber interface

units. For availability performance and capacity reasons each subscriber interface unit might serve e.g. up-to 1000 subscribers (this is purely hypothetical, each vendor has its own dimensioning rules). So, a local switch serving 100000 subscribers would have something like 100 subscriber interface units. Each of them would use some number of internal PCM-lines for connections to the switching fabric. E.g. if each subscriber interface unit uses 4 internal PCM –lines, the total takes 400 PCM –lines from the switching fabric interface. If the control computer capacities are such that 80 (PC-like) control computers are needed and each would be connected to the switching fabric also by 4 PCM-lines, this would consume 320 internal PCM –lines. As a result, still more than 7000 PCM lines would be available for connections to other equipment and to trunks towards neighboring exchanges.

Control part in Figure 1.2 consists of the control computers taking care of call signaling on the incoming and outgoing sides and also doing call control processing. Most often for a single telephone call one control computer would be dealing with the incoming signaling and incoming call control while another control computer would take care of outgoing call control and outgoing signaling. On both incoming and outgoing sides, the particular control computer would be decided based on the incoming circuit identity and

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S38.3115 Signaling Protocols – Lecture Notes outgoing circuit identity – recall that we assume that the mapping of trunks and internal PCM lines that carry user’s calls to control computers is semipermanent.

Like we discussed under call state, processing signaling and call control for a single call creates several state variables. These would reside in the respective control computers. During the process of setting up a call, the call progresses through many states reflecting the state of the communication on the incoming side, state of the switching fabric, state of the internal communications between different control computers, state of the outgoing call control and state of the outgoing signaling. The state of the switching fabric is replicated into memory so that it is easy to find free circuits. Moreover, reservation of a particular circuit from the fabric is recorded into a particular call’s state information.

A typical software bug in an exchange is such that some resource is reserved for a call and never released. If the bug persists, at some point all the resources are exhausted and the system ends up in a deadlock. To avoid that, for example, we can set a time limit for any state. Such a principle would mean that the system will always strive towards an idle state even if explicit release of some resource does not happen.

For the purpose of controlling call traffic an exchange needs quite a lot of information. It is typical that an exchange has a memory resident database. In particular, a local exchange has a memory resident subscriber database. The exchange database stores information about the users and the topology of the network around the node. An exchange may also need to collect charging data about calls. Moreover, an exchange usually needs to be connected to a network management system, so that the node can be managed in terms of failures, configuration, accounting, performance and security (FCAPS).

Fault-tolerance requirements on switching systems

It is required in ITU-T standards and by operators that a switching system has a downtime of less than about 2 min/year (we talk about 5 nines availability performance = the probability of failure is < 0.99999).

Even a single full restart of a distributed system can easily take longer. It follows that spontaneous restarts because of hardware or software faults need to be avoided for several years of operations on average. It also follows that many equipment blocks need to be duplicated or at least some level of hardware redundancy need to be provided. Typically, the switching fabric is duplicated. For example in DX 200, control computers that deal with signaling support either duplication or the so called N+1 –redundancy schema.

To make use of the hardware replication, signaling and call control

computations need to be replicated as well. If a control computer is duplicated, lecture 1

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S38.3115 Signaling Protocols – Lecture Notes the active machine takes care of the signaling and call control load while the

hot standby computer mirrors the state of the active one but is not visible to the outside world. In case, the active machine fails, the recovery system of the exchange orchestrates the switch over or fail-over of the duplicated machine.

The target is that as few of the ongoing calls as possible see the difference because of the fail-over. This is achievable because of the mirroring of call and signaling state in the standby machine that takes over the load after the failover. However, it would be unfeasible to store the call states of calls handled by a control computer to a disk and restore the state after failover to the new machine. While the system would be retrieving data from disk, some event might happen in a neighboring exchange that our node under recovery would miss. Therefore, call state information is normally not stored on disk. Instead, call and signaling state is maintained only in random access memory.

The particular replication schema for equipment in an exchange is decided based on the impact of failure of the particular equipment. If for example less than 100 subscribers would be affected, probably no replication is needed. If the failure of a part would lead to full system restart or downtime of the whole system, no less than full duplication will suffice.

What happens to switch function if calls are carried over an

IP network

Like we discussed, signaling is still needed. Also a limited set of call control

(or session control) actions are needed. These functions can be easily run on, for example, a Linux based control computer with Ethernet connectivity. Such computers are called for example call managers, call session controllers etc.

Such a computer does not need to control a switching fabric. Instead voice packets are carried over the IP network and packet forwarding takes place in routers under the direction of IP-routing packet by packet. It may still be necessary for a session controller to control Media Gateways and some middle boxes such as firewalls.

In VOIP, voice is encoded into packets in the end devices and decoded back to voice also in the end devices. If the connection is over a packet network end to end, the network does not need to understand anything about voice coding. It is up to the end devices to agree on the use of any codec, standard or proprietary. Because codecs are just software on general purpose digital hardware and the cost of that software and hardware is born by the user, it has become feasible to use many codecs.

Generalization: telephony paradigm

Let us take a high level view of the digital transmission based on PCM and digital switching from the users and operators point of view. This leads us to lecture 1

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S38.3115 Signaling Protocols – Lecture Notes formulate the networking paradigm that is the foundation of telephony networks.

We can say that in telephony networks on user request the operator establishes a (64kbit/s) channel connecting the user to another counting the time for using the channel. On user request the operator releases the channel and stops counting. What the two communicating users do with the channel it is up to them. Based on the counted time, in each billing period, the operator sends a bill to the user.

For example it does not matter that half of the time during a voice call each user is silent listening what the other is saying. The channel is still reserved in both directions.

From this paradigm it follows that keeping track of the channels is important.

It is the task of call control. The networking paradigm in IP networks is quite different and the economics are quite different as well. When we move to voice over IP, it will be interesting to see how the paradigm shift will impact telephony services.

Differences between public and private switched networks

In public networks such as the PSTN, we call the switching nodes exchanges

(puhelinkeskus). In private (or corporate) networks we call similar nodes

Private (Automatic) Branch Exchanges or PABXs (PBX) (vaihde in Finnish).

Although the functions of exchanges and PABXs are similar, there are significant differences. One should also note that the products used in public and private network are of different brands. The differences can be explained by the differences in requirements for private and public networking.

For example

• Maximum capacities of public switching systems are large, smaller nodes are enough for private networks,

• Large capacity leads most often to a distributed design while in a

PABX even a single computer may be enough to handle all signaling and call control load.

• Availability performance requirements for the node and for many parts are stricter for public networks than for private networks. (We discussed earlier the famous down-time requirement of 2 min/year for public network switches). In a public exchange many subsystems such as the switching fabric, many types of control computers are duplicated or at least some replication schema is used to improve availability performance. In PABXs this may not be necessary to the same extent.

• Wired subscriber line requirements are easier for PABXs than for public subscriber exchanges (cmp indoor installations to outdoor). lecture 1

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S38.3115 Signaling Protocols – Lecture Notes

• A very wide set of services (supplementary services) are provided by

PABXs, usually a poorer set is enough for public networks.

• International or at least national standards for signaling are typical of public networks while proprietary variants of common standards are typical in PABXs.

• Charging is very important in Public networks, in PABXs call statistics may be collected but usually it is used for company internal purposes if at all in the corporation whose network is supported by the PABX.

Duplicating statistics for outgoing calls in PABX (and the local switch) may also be provided and used but usually call charges are just a minor item in company costs while for the operator accurate call charge data collection is a matter of life and death.

• Used signaling systems are different. PABXs use private network signaling systems while public exchanges mostly support public network signaling systems. A common denominator is also needed – otherwise the two could not talk to each other. Public exchanges may support international signaling and inter-operator signaling systems, these do not appear on PABXs.

• Both are moving towards voice over IP, PABXs are leading the way.

Internet Architecture for Voice over IP

The original Internet architecture followed the end-to-end principle as formulated by Dave Clark in 1984. This principle says that we should not

place into the network any function that can not completely be implemented in the network. Instead, such functions should be left to be implemented in end

devices or hosts. It followed from this principle, for example, that communication error correction is taken care of by TCP in the hosts while the network concentrates on routing packets from one host to another. In the original Internet also all hosts could directly see each other because each one of them had a static IP address. For the hosts, the network is transparent but if they wish, they can find out its structure by for example using the ping or traceroute utility.

All this is the idealized past. Today’s reality of IP networks is different.

Corporate and many residential users are behind NATs, so they use private IP addresses. Most users have dynamic IP addresses that are allocated at host startup or even later.

Most corporate users are behind Firewalls. These may normally let packets pass from the protected network to the public Internet and take care of letting the acknowledgements also come through. If however, someone wishes to take the initiative and first send a packet from the public network to a host that sits behind a Firewall, there is a problem in such packet pushing. The firewall is likely to block the pushed packet. This means that telephony that requires sending signals from the originating A-subscriber to the callee or B-subscriber lecture 1

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S38.3115 Signaling Protocols – Lecture Notes is far from straightforward in modern IP networks. We will discuss NAT traversal in one of the later lectures in detail.

To recognize the existence of NATs, Firewalls and other middle-boxes, Dave

Clark formulated the Trust-to-Trust principle in 2007. This says that

“The function in question can completely and correctly be implemented only with the knowledge and help of the application standing at points where it can be trusted to perform its job properly.” —

David Clark, MIT Communications Futures Program, Bi-annual meeting, May 30-31, 2007,

Philadelphia, PA.

The reality of IP networks in a corporate environment is such that every stakeholder wishes to hide its network from every other stakeholder. For this purpose they place session border controllers or application gateways on administrative boundaries. An application gateway processes all packets that go through it on application level. For telephone calls it will process call signaling and also the voice packets. The result is that a call may need to traverse e.g. 3…9 networks that are invisible to each other, traverse application gateways on the borders of those networks and also that end to end delay grows to unacceptable levels. It remains to be seen how these problems will be resolved.

Network intelligence vs. terminal intelligence

We speak loosely about intelligence referring to the software and its role for service implementation. Traditional PSTN networks assume no software in phones, rather for a long time, PSTN phones used to be electro-mechanical devices. Sometimes, we label PSTN phones dumb.

The Internet, on the other hand, was from the very beginning built with the idea that the network is connecting computers and that the computers implement the services while the role of the network is just to transfer packet i.e. provide a bit-pipe. Two other major example networks that we will talk about are ISDN and GSM/3G networks. In the allocation of “intelligence” these networks reside somewhere between PSTN and the Internet. We will see on the ISDN lecture that ISDN actually makes poor use of the idea that terminals are computers although in practice they must be so. User devices in mobile networks such as GSM and 3G are battery powered, so power consumption of mobiles has always played a major role in the design of cellular systems. Nevertheless, one can claim that cellular systems make a fairly good use of the idea that mobiles are “intelligent” without going as far as the Internet.

The historical trend in communication networks over the past several tens of years has been that “intelligence” is moving from the network to the user devices. One should expect additional moves in this direction in the future.

Fundamentally, why is this happening? We point out a few reasons: lecture 1

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S38.3115 Signaling Protocols – Lecture Notes

• User devices are created under consumer market economies. Volumes drive the prices down. As a result a lot of more value to the consumer can be created in the terminal. On the other hand, network equipment volumes are several orders of magnitude lower.

• Users love to own gadgets. Branding of the gadgets leads to users being willing to pay several 100€ for their device. At the current and future level of electronics, a lot of functionality can be realized for that money provided volumes are high.

• Intelligent user devices facilitate service innovation by the users themselves. Of course users know best what they need. No vendor or operator can compete in this understanding.

• Networks that allow and rely on intelligent user devices assign communication services value to the users – the users benefit.

Networks that try to control accurately what the users can do aim to draw as much of the value as possible to the operator of the network. If the users have a choice they will choose to use dumb networks and intelligent terminals.

Scope of our discussion on this course

Switching and signaling form the infrastructure on top which communication services are implemented. Communication networking is one of the largest industries in the world. Networks are supposed to create social value, a part of which is realized in operator revenues and equipment vendor revenues. New networks require huge investments.

In the era of digital communication networks, many networking systems have been created that have under-performed on the market compared to high expectations that were originally placed on them. This can not be explained only by relative technical merits of the systems. Also, the roles of consumers, corporations and operators are important in shaping the technical solutions and the outcomes on the market.

For analyzing the economics of different system we will not create a framework on this course – these subjects are discussed in depth on other courses given by ComNet.

Nevertheless, with this historical background when describing different signaling and networking solutions, we will discuss not only technical merits of the systems but also point out some economic reasoning behind the fundamental design decisions.

On this course we will start from systems that historically resided on the boundary of the move from analogue telephony to digital telephony and move through different digital telephony systems to packet based telephony that is currently under development and early deployment. In explaining the lecture 1

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S38.3115 Signaling Protocols – Lecture Notes technology we will try to more or less consistently pay attention to a set of issues related to each new signaling system. On the lecture on ISUP, after first describing a few example signaling systems we will make a first attempt of drawing a picture of this mental framework.

We can summarize the goal of the course in the following way:

Signaling will be analyzed on a functional level. Focus is on understanding advantages, drawbacks and fundamental design aspects of widespread solutions in different networks. Through understanding several signaling systems, we consider how such systems interwork. By comparing circuit and packet networks we build understanding of the technology trends and how functionality from legacy networks is inherited into new systems. lecture 1

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