VOIP

Voice Compression and Coding

 Speech Codecs
• Waveform codec
• Source codec
(vocoders)
• Hybrid codec
 Waveform Codec
• Waveform codec’s attempt, without using any
knowledge of how the signal to be coded was
generated, to produce a reconstructed signal whose
waveform is as close as possible to the original.
• This means that in theory they should be signal
independent and work well with non-speech signals.
• Generally they are low complexity codec’s which
produce high quality speech at rates above about 16
kbits/s.
• When the data rate is lowered below this level the
reconstructed speech quality that can be obtained
degrades rapidly
 Source Codec
• Source coders operate using a model of how the source was
generated, and attempt to extract, from the signal being
coded, the parameters of the model.
• It is these model parameters which are transmitted to the
decoder.
• Source coders for speech are called vocoders, and work as
follows.
• The vocal tract is represented as a time-varying filter and is
excited with either a white noise source, for unvoiced speech
segments, or a train of pulses separated by the pitch period
for voiced speech.
• Therefore the information which must be sent to the decoder
is the filter specification, a voiced/unvoiced flag, the
necessary variance of the excitation signal, and the pitch
period for voiced speech.
 Hybrid Codec
• Hybrid codecs attempt to fill the gap between
waveform and source codecs.
• Waveform coders are capable of providing good quality
speech at bit rates down to about 16 kbits/s, but are of
limited use at rates below this.
• Source coders on the other hand can provide
intelligible speech at 2.4 kbits/s and below, but cannot
provide natural sounding speech at any bit rate.
• Although other forms of hybrid codecs exist, the most
successful and commonly used are time domain
Analysis-by-Synthesis (AbS) codecs.
 PCM - G.711
• Pulse Code Modulation (PCM) codecs are the simplest form
of waveform codecs.
• Narrowband speech is typically sampled 8000 times per
second, and then each speech sample must be quantized.
• If linear quantization is used then about 12 bits per sample
are needed, giving a bit rate of about 96 kbits/s.
• However this can be easily reduced by using non-linear
quantization.
• For coding speech it was found that with non-linear
quantization 8 bits per sample was sufficient for speech
quality which is almost indistinguishable from the original.
• This gives a bit rate of 64 kbits/s, and two such non-linear
PCM codecs were standardised in the 1960s
 Adaptive Differential PCM (ADPCM)
• Adaptive Differential Pulse Code Modulation (ADPCM)
codecs are waveform codecs which instead of
quantizing the speech signal directly, quantize the
difference between the speech signal and a prediction
that has been made of the speech signal.
• If the prediction is accurate then the difference
between the real and predicted speech samples will
have a lower variance than the real speech samples,
and will be accurately quantized with fewer bits than
would be needed to quantize the original speech
samples.
 DPCM - G.721 , G.726 & G.727
• In the mid 1980s the CCITT standardised a 32
kbits/s ADPCM, known as G721, which gave
reconstructed speech almost as good as the
64 kbits/s PCM codecs.
• Later in recommendations G726 and G727
codecs operating at 40,32,24 and 16 kbits/s
were standardised.
 Code-Excited Linear Predictive (CELP)
• At bit rates of around 16 kbits/s and lower the
quality of waveform codecs falls rapidly, as can be
seen in figure shown earlier.
• Thus at these rates hybrid codecs, especially CELP
codecs and their derivatives, tend to be used.
• However because of the forward adaptive
determination of the short term filter coefficients
used in most of these codecs, they tend to have
high delays.
 G.728 (Low-Delay) CELP Codecs
• CELP codec which was developed at AT&T Bell
Labs, and was standardised in 1992 as G728.
• This codec uses backward adaption to calculate
the short term filter coefficients, which means
that rather than buffer 20 ms or so of the input
speech to calculate the filter coefficients they are
found from the past reconstructed speech.
• This means that the codec can use a much
shorter frame length than traditional CELP
codecs, and G728 uses a frame length of only 5
samples giving it a total delay of less than 2 ms.
 G.723 (Algebraic Code-Excited Linear
Prediction (ACELP)
• Normal conversation involves significant periods of
silence.
• G723 specifies a mechanism for silence suppression
where Silence Insertion Description (SID) frames can
be used.
• These are only 32bits long – this means that silence
only occupies 1Kbps – compared to 64Kbps for
G711.
• G.723 has an MOS score of 3.8 but has a delay of
37.5 mSecs at the encoder
 G.729
• G.729 is an umbrella of vocoder standards.
• The G.729 codec perform voice compression at
bit rates that vary between 6.4 and 12.4 kbps.
• The figure below shows an example of the G.729
vocoder connected to a digital communication
channel.
• The input speech is fed into the G.729 encoder as
a stream of 16-bit PCM samples, sampled at a
rate of 8000 samples/second.
• The G.729 encoder compresses the data into the
Encode Stream.
 G.729 ....
• G.729 also uses samples of the actual human
speech to set the vocoder settings properly.
• It also compares the actual voice from the
synthetic voice to come up with a "code."
• The code along with the vocoder settings are
what's sent to the remote end.
• The remote end takes the code and vocoder
settings and plays the sound.
RTP
 RTP
• Ver. This 2-bit field defines the version number. The current version is 2.
• P. This 1-bit field, if set to 1, indicates the presence of padding at the end
of the packet. In this case, the value of the last byte in the padding
defines the length of the padding. Padding is the norm if a packet is
encrypted. There is no padding if the value of the P field is 0.
• X. This 1-bit field, if set to 1, indicates an extra extension header
between the basic header and the data. There is no extra extension
header if the value of this field is 0.
• Contributor count. This 4-bit field indicates the number of contributors.
Note that we can have a maximum of 15 contributors because a 4-bit
field only allows a number between 0 and 15.
• M. This 1-bit field is a marker used by the application to indicate, for
example, the end of its data.
• Payload type. This 7-bit field indicates the type of the payload. Several
payload types have been defined so far.
RTP
 RTP
• Sequence number. This field is 16 bits in length. It is used to number the RTP
packets. The sequence number of the first packet is chosen randomly; it is
incremented by 1 for each subsequent packet. The sequence number is used by
the receiver to detect lost or out of order packets.
• Timestamp. This is a 32-bit field that indicates the time relationship between
packets. The timestamp for the first packet is a random number. For each
succeeding packet, the value is the sum of the preceding timestamp plus the time
the first byte is produced (sampled). The value of the clock tick depends on the
application. For example, audio applications normally generate chunks of 160
bytes; the clock tick for this application is 160. The timestamp for this application
increases 160 for each RTP packet.
• Synchronization source identifier. If there is only one source, this 32-bit field
defines the source. However, if there are several sources, the mixer is the
synchronization source and the other sources are contributors. The value of the
source identifier is a random number chosen by the source. The protocol provides
a strategy in case of conflict (two sources start with the same sequence number).
• Contributor identifier. Each of these 32-bit identifiers (a maximum of 15) defines
a source. When there is more than one source in a session, the mixer is the
synchronization source and the remaining sources are the contributors.
 RTP
• Although RTP is itself a transport layer protocol, the RTP
packet is not encapsulated directly in an IP datagram.
• Instead, RTP is treated like an application program and is
encapsulated in a UDP user datagram.
• However, unlike other application programs, no well-
known port is assigned to RTP.
• The port can be selected on demand with only one
restriction: The port number must be an even number.
• The next number (an odd number) is used by the
companion of RTP, Real-Time Transport Control Protocol
• (RTCP).
 RTCP
• RTP allows only one type of message, one that
carries data from the source to the destination.
• In many cases, there is a need for other messages
in a session.
• These messages control the flow and quality of
data and allow the recipient to send feedback to
the source or sources.
• Real-Time Transport Control Protocol (RTCP) is a
protocol designed for this purpose. RTCP has five
types of messages,
 RTCP
• Sender Report
• The sender report is sent periodically by the active senders in
a conference to report transmission and reception statistics
for all RTP packets sent during the interval.
• The sender report includes an absolute timestamp, which is
the number of seconds elapsed since midnight January 1,
1970.
• The absolute timestamp allows the receiver to synchronize
different RTP messages.
• It is particularly important when both audio and video are
transmitted (audio and video transmissions use separate
relative timestamps).
 RTCP
Receiver Report
• The receiver report is for passive participants, those that do not send RTP
packets.
• The report informs the sender and other receivers about the quality of
service.
Source Description Message
• The source periodically sends a source description message to give
additional information about itself.
• This information can be the name, e-mail address, telephone number, and
address of the owner or controller of the source.
Bye Message
• A source sends a bye message to shut down a stream. It allows the source to
announce that it is leaving the conference.
• Although other sources can detect the absence of a source, this message is a
direct announcement. It is also very useful to a mixer
 RTCP
Application-Specific Message
• The application-specific message is a packet for an
application that wants to use new applications (not
defined in the standard).
• It allows the definition of a new message type.
UDP Port
• RTCP, like RTP, does not use a well-known UDP port.
• It uses a temporary port.
• The UDP port chosen must be the number immediately
following the UDP port selected for RTP, which makes it
an odd-numbered port.
VOIP signalling protocols
 H.323
Operation
• Let us show the operation of a telephone communication using H.323 with a
simple example.
• The steps used by a terminal to communicate with a telephone.
1. The terminal sends a broadcast message to the gatekeeper. The gatekeeper
responds with its IP address.
2. The terminal and gatekeeper communicate, using H.225 to negotiate
bandwidth.
3. The terminal, the gatekeeper, gateway, and the telephone communicate using
Q.931 to set up a connection.
4. The terminal, the gatekeeper, gateway, and the telephone communicate using
H.245 to negotiate the compression method.
5. The terminal, gateway, and the telephone exchange audio using RTP under the
management of RTCP.
6. The terminal, the gatekeeper, gateway, and the telephone communicate using
Q.931 to terminate the communication.
H.323
 QoS in VoIP
• QoS (Quality of Service) is a major issue in VOIP implementations. The
issue is how to guarantee that packet traffic for a voice or other media
connection will not be delayed or dropped due interference from other
lower priority traffic.
• Things to consider are
•
– Choice of codec
– Latency: Delay for packet delivery
– Jitter: Variations in delay of packet delivery
– Packet loss: Too much traffic in the network causes the network to drop
packets
– Burstiness of Loss and Jitter: Loss and Discards (due to jitter) tend to occur in
bursts
– Call set up time
– Call success ratio
– Call set up rate (Calls /Sec)
 QoS in VoIP
• Quality of service (QoS) is an internetworking
issue that has been discussed more than
defined.
• We can informally define quality of service as
something a flow of data seeks to attain.
• Although QoS can be applied to both textual
data and multimedia, it is more an issue when
we are dealing with multimedia.
 QoS in VoIP
• Flow Characteristics
• Traditionally, four types of characteristics are attributed to a flow: reliability, delay,
jitter, and bandwidth.
Reliability
• Reliability is a characteristic that a flow needs.
• Lack of reliability means losing a packet or acknowledgment, which entails
retransmission. However, the sensitivity of application programs to reliability is not
the same.
• For example, it is more important that electronic mail, file transfer, and Internet
access have reliable transmissions than
• telephony or audio conferencing.
Delay
• Source-to-destination delay is another flow characteristic.
• Again applications can tolerate delay in different degrees. In this case, telephony,
audio conferencing, video conferencing, and remote log-in need minimum delay,
while delay in file transfer or e-mail is less important.
 QoS in VoIP
Jitter
• Jitter is the variation in delay for packets belonging to the same flow.
• For example, if four packets depart at times 0, 1, 2, 3 and arrive at 20, 21, 22,
23, all have the same delay, 20 units of time.
• On the other hand, if the above four packets arrive at 21, 23, 21, and 28,
they will have different delays: 21, 22, 19, and 24.
• For applications such as audio and video, the first case is completely
acceptable; the second case is not.
• For these applications, it does not matter if the packets arrive with a short or
long delay as long as the delay is the same for all packets.
• For this application, the second case is not acceptable.
• Jitter is defined as the variation in the packet delay.
• High jitter means the difference between delays is large; low jitter means the
variation is small.
Bandwidth
• Different applications need different bandwidths. In video conferencing we
need to send millions of bits per second to refresh a color screen while the
total number of bits in an e-mail may not reach even a million.
 QoS in VoIP
• Flow Classes
• Based on the flow characteristics, we can classify
flows into groups, with each group having similar
levels of characteristics.
• This categorization is not formal or universal;
some protocols such as ATM have defined
classes.
Techniques to Improve QoS
scheduling, traffic shaping, admission control, and
resource reservation.
MPLS
RSVP
 Introduction of RSVP
• Resource ReSerVation Protocol.
• Allows applications running in hosts to reserve
resources in the Internet for their data flows.
• Used by the routers to forward bandwidth
reservation requests.
• RSVP software must be present in the receivers,
sender, and routers.
 Introduction of RSVP (cont.)
• Two principle characteristics of RSVP
– It provides reservations for bandwidth in multicast
trees(unicast is handled as a special case).
– It is receiver-oriented.
• RSVP reserves resources for only one direction data
streams.
• RSVP is not a routing protocol
– It does not determine the links in which the reservations are
to be made.
– An RSVP daemon consults the local routing databases to
obtain routes.
 Introduction of RSVP (cont.)
• RSVP depends on an underlying routing
protocol(unicast or multicast) to determine the
routes for the flows
• RSVP is sometimes referred to as a signaling
protocol that allows hosts to establish and tear-
down reservations for data flows
RSVP: multicast- and receiver-oriented.
 Heterogeneous receivers
• Sender does not have to know the receiving rates
of all receivers.
• It only needs to know the maximum rate of all its
receivers.
• The sender encodes the video or audio into
multiple layers and sends all the layers up to the
maximum rate into multicast tree.
• The receivers pick out the layers that are
appropriate for their receiving rates.
 Heterogeneous receivers (cont.)
• In order to not excessively waste bandwidth in
the network’s links, the heterogeneous receivers
must communicate to the network the rates they
can handle.
• RSVP gives foremost attention to the issue of
reserving resources for heterogeneous receivers.
RSVP Operation Example

Path message Session

(Ipa,PID,Port)
Resv message
IGMP(1)
IGMP message
Receiver B
Data Resv (3)
Packet
Session
(Ipa,PID,Port) (4)

path (2) IGMP (1)

Sender

Resv(3) Receiver A
Merge Session
point (Ipa,PID,Port)
A Few Simple Examples

An RSVP example
 An RSVP video conference example

• Each router receives a reservation message from

each of its downstream links in the multicast tree
and sends only one reservation message into its
upstream link.
 Call Admission
• Whenever a router receives a new reservation
message, it must first determine if its
downstream links on the multicast tree can
accommodate the reservation.
• This admission test is performed whenever a
router receives a reservation message.
• RSVP does not define the admission test, but it
assumes that the routers perform such a test and
that RSVP can interact with the test.
 Path Messages
• Path messages are another important RSVP message type.
• Originate at the senders and flow downstream towards the
receivers.
• The principle purpose of the path messages is to let the routers
know on which links they should forward the reservation
messages.
• The path messages also contain a sender Tspec, which defines
the traffic characteristics of the data stream that the sender will
generate.
• Tspec can be used to prevent over reservation.
 Resv messages
• After a receiver has received a Path message, it sends a Resv
message.
• The Resv message travels toward the sender (upstream) and
makes a resource reservation on the routers that support
RSVP.
• If a router does not support RSVP on the path, it routes the
packet based on the best-effort delivery methods.
 Reservation merging
• In RSVP, the resources are not reserved for each receiver in a flow; the
reservation is merged. In Figure, Rc3 requests a 2-Mbps bandwidth while Rc2
requests a 1-Mbps bandwidth.
• Router R3, which needs to make a bandwidth reservation, merges the two
requests. The reservation is made for 2 Mbps, the larger of the two, because a 2-
Mbps input reservation can handle both requests.
• The same situation is true for R2. Rc2 and Rc3, both belonging to one single flow,
request different amounts of bandwidth.
• In a multimedia environment, different receivers may handle different grades of
quality.
• For example, Rc2 may be able to receive video only at 1 Mbps (lower quality),
while Rc3 may be able to receive video at 2 Mbps (higher quality).
 Reservation Styles
• A reservation message specifies whether merging
of reservations from the same session is
permissible.
• A reservation style also specifies from which
senders in a session the receiver desires to
receive data.
• There are currently three reservation styles
– Wildcard-filter style.
– Fixed-filter style.
– Shared-explicit style.
 Reservation Styles (cont.)
• Wildcard-Filter Style
– It is telling the network that it wants to receive all flows
from all upstream senders in the session and that its
bandwidth reservation is to be shared among the
senders.
• Fixed-Filter Style
– It specifies a list of senders from which it wants to
receive a data flow along with a single bandwidth
reservation. These reservation are distinct, i.e., they
are not to be shared.
 Reservation Styles (cont.)
• Shared-Explicit Style
– It specifies a list of senders from which it wants to
receive a data flow along with a single bandwidth
reservation. This reservation is to be shared among all
the senders in the list.
Reservation Styles (cont.)
• Shared reservations, created by the wildcard-filter
and the shared-explicit styles, are appropriate for
a multicast session whose sources are unlikely to
transmit simultaneously .
• The fixed-filter reservation, which creates distinct
reservations for the flows from different senders,
is appropriate for video teleconferencing.
Examples of Reservation Styles

Sample scenario for RSVP reservation styles

Wildcard filter reservations.

fixed filter reservations

shared-explicit reservations
 Soft State
• The reservation in the routers and hosts are
maintained with soft states.
• Each reservation for bandwidth stored in a router
has an associated timer.
• If a receiver desires to maintain a reservation, it
must periodically refresh the reservation by
sending reservation messages.
• A receiver can also change its reservation by
adjusting its reservation in its stream of refresh
messages.
 Soft State (cont.)
• The senders must also refresh the path state by
periodically sending path messages.
 Transport of Reservation Messages
• RSVP messages are sent hop-by-hop directly over
IP, thus the RSVP message is placed in the
information field of the IP datagram.
• If an RSVP path or reservation message is lost, a
replacement refresh message should arrive soon.
 Disadvantage of RSVP
• Need more memory to record per flow state
information of each node in network.
• RSVP is lack of scalability.
Wireless in Local Loop
 Wireless in Local Loop
• What is WLL?
- WLL is a system that connects subscribers
to the local telephone station wirelessly.
• WLL systems can be based on one of the four
below technologies:
– Satellite-based systems.
– Cellular-based systems.
– Microcellular-based Systems
– Fixed Wireless Access Systems
A general WLL setup
 WLL Architecture
UWLL

WANU
Transceiver WASU

Air TWLL
Trunk Switch WLL AM Interface
PSTN
function Controller HLR

Wireless Access Network Unit(WANU) Wireless Access Subscriber Unit(WASU)

– Interface between underlying telephone – located at the subscriber
network and wireless link
– translates wireless link into a
– consists of traditional telephone connection
• Base Station Transceivers (BTS)
• Radio Controller(RPCU)
• Access Manager(AM)
• Home Location Register(HLR)
 WLL Architecture
The given architecture consists of three major components i.e WANU, WASU and SF
• Wireless Access network unit (WANU): the WANU consists of various components which
include
• several base stations transceivers or radio ports (RP)
• Radio port control unit
• an Access manager (AM)
• an HLR.
• It provides various functionalities like:
Authentication
Air interface privacy
Over-the-air registration of subscriber units.
Operations and Maintenance
Routing
Billing
Switching functions
Transcoding of voice and data.
 WLL Architecture

Wireless access subscriber unit (WASU): It provides an air interface UWLL towards
the network and a traditional interface TWLL towards the subscriber.

• The power supply for it is provided locally.

• The interface includes
protocol conversion and transcoding
authentication functions
signaling functions
• The TWLL interface can be an RJ-11 or RJ-45 port.
• The UWLL interface can be AMPS, GSM, DECT and so one.

• Switching Function (SF): The switching function (SF) is associated with a switch
that can be digital switch with or without Advanced Intelligent Network (AIN)
capability, an ISDN switch or a Mobile Switching Centre (MSC).
• The AWLL interface between the WANU and the SF can be ISDN-BRI or IS-634 or IS-
653 or such variants.
IP over ATM
 IP over ATM
 Although intended to be ubiquitous, the ATM-only
dream was not realized
 ATM co-exists with many legacy networks
 legacy networks: Ethernet, Token Ring etc.
 IP remains a popular internetworking solution
 Therefore, it is important to know how IP packets
can be transmitted over ATM networks
 IP over ATM Architectures

 IP machines (desktop or router) connected to

ATM networks need mechanisms to run
connectionless IP over connection-oriented ATM
 Two standard mechanisms
 LAN Emulation
 Classical IP over ATM
 LAN Emulation (LANE)
LANE : Motivations
 Quick deployment of ATM
 Makes ATM look like either Ethernet or Token
Ring
 No changes required for existing network
software
 Existing network protocols and software, e.g. IP,
IPX, AppleTalk, running on Ethernet will also run
on ATM
 LAN Emulation (LANE)
LANE Address Structure
 Each host connected to a LANE
network has three addresses
IP Address
 IP address : needed for
internetworking
 MAC address: an Ethernet or Token
Ring address to emulate an Ethernet Ethernet Address
or a Token Ring
 ATM address : physical ATM address
to locate devices in ATM network
ATM Address
 LAN Emulation (LANE)

 Each host runs a LAN Emulation Client (LEC); it

 does the address resolution (MAC-ATM) for the host
 sets up data connections to other hosts
 sets up control connections to LANE servers
 Each host communicates with 3 servers
 Broadcast and Unknown Server (BUS)
 LAN Emulation Server (LES)
 LAN Emulation Configuration Server (LECS)
 LAN Emulation (LANE)
LANE Servers

LECS
LEC LEC
LES

BUS
LEC LEC
 LAN Emulation (LANE)
What LANE Servers Do
Server Functions
LECS Provides LES's ATM address and the MAC type details to
the LEC
LES Does MAC-ATM address resolution
BUS When a host wants to broadcast a packet, it sends it to
BUS, the BUS then forwards it all the hosts connected to
the Emulated LAN (ELAN)

IP-MAC address resolution is done by ARP servers as

usual. On top of that, MAC-ATM resolution is needed.
 LAN Emulation (LANE)
LANE Sequence of Events

Discover LECS’s ATM address and establish

(LECS) Initialization communication with the LECS

Based upon some policy, the LECS assigns the client to a

Configuration particular LES (gives the LES’s ATM address to client)

The client joins the ELAN by registering its ATM and MAC
Registration with LES address with the LES

The client obtains the BUS’s ATM address from LES. This
(BUS) Initialization address is used for broadcasting.

The client now can send MAC frames. Address resolution

Data Transfer with LES may be necessary. ATM VC setup may be req’d.
 Interoperability
Legacy LAN and LANE

Ethernet LAN ATM-LAN ATM LANE

Converter

Physical Scenario

Extended Ethernet

Logical Scenario
 Classical IP over ATM (CLIP)

 LANE hides ATM technology and mimics

Ethernet
 Since LANE mimics Ethernet, it inherits
Ethernet’s undesirable features
 broadcast-like competition for resources
 No ATM QoS available to network layer
 Classical IP over ATM (CLIP)
CLIP Basics

 CLIP treats ATM as a new link layer network

technology
 Advantages : With CLIP, the host software is
aware of ATM and makes use of ATM QoS
 Disadvantages
 Ethernet-based software will not work
 CLIP is IP-only solution, (does not cover IPX etc.)
 Classical IP over ATM (CLIP)
LANE vs CLIP

Existing IP Existing IPX Existing Netbeui

LANE

Supports all existing software

New IP CLIP
Cannot support any existing software,
supports only new IP software
 Classical IP over ATM (CLIP)
CLIP Components

 There is only one server, called ATM ARP

Server
 Instead of using broadcasting, the IP host
communicates with this server directly
whenever it needs to resolve IP-ATM
addresses
Classical IP over ATM (CLIP)

Classical IP Over ATM

Connecting Hosts with ARP Server

Client ATM ARP ATM ARP Server

Request
Reply to ARP
Request

ATM Network
Client

Client
 Classical IP over ATM (CLIP)
CLIP Sequence of Events

When an IP host joins a LAN, it registers its ATM and IP

Registration with ARP Server addresses with the ATM ARP server

When an IP host wants to communicate with another, it

Obtain ATM addr of dest obtains the ATM address of the dest from ARP server if it
is not in the local cache

The client sets up an ATM VC (if such a VC is not there)

Data Transfer with the destination and sends ATM cells over the VC.
SONET/SDH
 Idea behind SONET
Synchronous Optical NETwork
• Designed for optical transport (high bitrate)
• Direct mapping of lower levels into higher ones
• Carry all PDH types in one universal hierarchy
– ITU version = Synchronous Digital Hierarchy
– different terminology but interoperable
• Overhead doesn’t increase with rate
• OAM designed-in from beginning
 SONET/SDH architecture

Layers
SONET was designed with definite layering concepts
Physical layer – optical fiber (linear or ring)
– when exceed fiber reach – regenerators
– regenerators are not mere amplifiers,
– regenerators use their own overhead
– fiber between regenerators called section (regenerator section)
Line layer – link between SONET muxes (Add/Drop Multiplexers)
– input and output at this level are Virtual Tributaries (VCs)
– actually 2 layers
• lower order VC (for low bitrate payloads)
• higher order VC (for high bitrate payloads)
Path layer – end-to-end path of client data (tributaries)
– client data (payload) may be
• PDH
• ATM
• packet data
 SONET/SDH architecture

ADM regenerator ADM

Path Line Section Line Path
Termination Termination Termination Termination Termination

path
line line line
section section section section

SONET (SDH) has at 3 layers:

• path – end-to-end data connection, muxes tributary signals path section
– there are STS paths + Virtual Tributary (VT) paths
• line – protected multiplexed SONET payload multiplex section
• section – physical link between adjacent elements regenerator section

Each layer has its own overhead to support needed functionality

SDH terminology
 SONET/SDH architecture

A SONET signal is called a Synchronous Transport Signal

The basic STS is STS-1, all others are multiples of it - STS-N
The (optical) physical layer signal corresponding to an STS-N is an OC-N

SONET Optical rate

STS-1 OC-1 51.84M
STS-3 OC-3 155.52M *3
STS-12 OC-12 622.080M *4
STS-48 OC-48 2488.32M *4
STS-192 OC-192 9953.28M *4
 SONET / SDH frame rate

framing

Synchronous Transfer Signals are bit-signals (OC are optical)

Like all TDM signals, there are framing bits at the beginning of the frame
However, it is convenient to draw SONET/SDH signals as rectangles
 SONET / SDH frame rate
SONET STS-1 frame
90 columns
framing
9 rows

Each STS-1 frame is 90 columns * 9 rows = 810 bytes

There are 8000 STS-1 frames per second
so each byte represents 64 kbps (each column is 576 kbps)
Thus the basic STS-1 rate is 51.840 Mbps
 SONET / SDH frame rate
SDH STM-1 frame
270 columns

…
9 rows

Synchronous Transport Modules are the bit-signals for SDH

Each STM-1 frame is 270 columns * 9 rows = 2430 bytes
There are 8000 STM-1 frames per second
Thus the basic STM-1 rate is 155.520 Mbps
3 times the STS-1 rate!
 SONET / SDH frame rate
SONET/SDH rates
SONET SDH columns rate
STS-1 90 51.84M
STS-3 STM-1 270 155.52M
STS-12 STM-4 1080 622.080M
STS-48 STM-16 4320 2488.32M
STS-192 STM-64 17280 9953.28M

STS-N has 90N columns STM-M corresponds to STS-N with N = 3M

SDH rates increase by factors of 4 each time
STS/STM signals can carry PDH tributaries, for example:
• STS-1 can carry 1 T3 or 28 T1s or 1 E3 or 21 E1s
• STM-1 can carry 3 E3s or 63 E1s or 3 T3s or 84 T1s
 SONET/SDH tributaries

SONET SDH T1 T3 E1 E3 E4
STS-1 28 1 21 1
STS-3 STM-1 84 3 63 3 1
STS-12 STM-4 336 12 252 12 4
STS-48 STM-16 1344 48 1008 48 16
STS-192 STM-64 5376 192 4032 192 64

E3 and T3 are carried as Higher Order Paths (HOPs)

E1 and T1 are carried as Lower Order Paths (LOPs)
(the numbers are for direct mapping)
 STS-1 frame structure
90 columns
3 rows
9 rows

Synchronous Payload Envelope

6 rows

Transport
Section overhead is 3 rows * 3 columns = 9 bytes = 576 kbps
Overhead
TOH
framing, performance monitoring, management
Line overhead is 6 rows * 3 columns = 18 bytes = 1152 kbps
protection switching, line maintenance, mux/concat, SPE pointer
SPE is 9 rows * 87 columns = 783 bytes = 50.112 Mbps
Similarly, STM-1 has 9 (different) columns of section+line overhead !
 STM-1 frame structure
270 columns

RSOH

…
MSOH

Section
Overhead
SOH
STM-1 has 9 (different) columns of transport overhead !
RS overhead is 3 rows * 9 columns
Pointer overhead is 1 row * 9 columns
MS overhead is 5 rows * 9 columns
SPE is 9 rows * 261 columns
 SONET Overhead
• Overhead bytes are used by SONET
equipment (e.g., switches) for exchange of
control and signalling information, and as a
low bandwidth data channel
• Three types of overhead bytes
– section
– line
– path
 SONET Overhead (Cont’d)
• Section overhead: 9 bytes per frame
– Includes two framing bytes, plus other control
information for maintenance and provisioning
• Line overhead: 18 bytes per frame
– Control info, plus 9 bytes for data channel
• Path overhead: variable size
– Payload type, path status, etc.
– Transmitted as part of payload itself (SPE)
 SONET Framing (Cont’d)
• The SPE in an STS-1 frame has sufficient
capacity to carry a DS-3 (45 Mbps)
• There are many other ways to “carve up” the
capacity of an STS-1 into smaller units used
by the telco’s
• These are called Virtual Tributaries (VT’s)
 SONET Framing (Cont’d)
• Examples of VT’s:
– VT 1.5: requires 3 columns of 9 bytes each,
corresponding to North American DS1 (T1)
standard (1.544 Mbps)
– VT 2: 4 columns, corresponds to European
standard for 2.048 Mbps
– VT 3: 6 columns (54 bytes) per frame,
corresponds to 3.088 Mbps
– VT 6: 12 columns, 6.312 Mbps
 STS-1 Framing Example
90 columns
...
...
9 ...
...
rows
...
...
...
...
...
Section and VT 1.5 VT 2
Line Overhead
 SONET Framing (Cont’d)
• A “VT group” is 9 rows x 12 columns
– Can conveniently repackage into four VT 1.5, or
three VT 2, or two VT 3, or one VT 6
• An STS-1 frame can hold 7 VT groups per
frame (84 columns), with 1 column for path
overhead, and 2 columns empty
 SONET Framing (Cont’d)
• Higher rate SONET signals are obtained by
interleaving N STS-1’s to form an STS-N
(e.g., STS-3 = 155 Mbps)
• STS-N has 9 rows, and N x 90 columns
• Interleaving is done byte by byte
 SONET and ATM
• If the entire STS-1 payload is to be used for ATM
transmission, then there is no need to use VT’s
at all
• The 53-byte ATM cells are simply packaged into
the SPE portion of the STS-1 frame, as they fit
• Cells may wrap across STS-1 overhead bytes, or
even STS-1 frame boundaries
• Overhead byte keeps track of where ATM cell
boundaries lie
 STS-1 ATM Example
90 columns
...
...
9 ...
...
rows
...
...
...
...
...
Section and
Line Overhead Start of ATM Cells
 SONET Topologies

 SONET topology can be a mesh, but most often it is a dual ring.

 Standard component of SONET ring is an ADM (Add/Drop
Multiplexer)
– Drop one incoming multiplexed stream and replace it with
another stream.
– Used to make up bi-directional line switching rings.
 SONET Topologies
(a) pre-SONET multiplexing

MUX DEMUX MUX DEMUX

remove insert
tributary tributary

(b) SONET Add-Drop multiplexing

MUX ADM DEMUX

remove insert
tributary tributary
 SONET Topologies

OC-3n
OC-3n

b 3 ADMs

c
OC-3n
physical loop net
 SONET Topologies
SONET/SDH need to be highly reliable
Down-time should be minimal (less than 50 msec)
So systems must repair themselves (no time for manual intervention)
Upon detection of a failure (dLOS, dLOF, high BER)
the network must reroute traffic (protection switching)
from working channel to protection channel
The Network Element that detects the failure (tail-end NE)
initiates the protection switching
The head-end NE must change forwarding or to send duplicate traffic
Protection switching is unidirectional
Protection switching may be revertive (automatically revert to working channel)

working channel

protection channel
head-end NE tail-end NE
 SONET Topologies

a a
ADM

d ADM ADM b d b

ADM

c c

(a) Dual ring (b) Loop-around in response to

fault
SONET Topologies

SONET Ring
 SONET Topologies

Regional Metro
Ring Ring Inter-Office
Rings