TrendCommunications

TrendCommunications+44 (0)1628 503500 +44 (0)1628 503500+33 1 69 35 54 70+49 (0) 89-32 30 09-30+1 256 461 0790+34 93 300 3313+91-22-28521059+86-10-8518-3141Infoline@trendcomms.comwww.trendcomms.com

International:United Kingdom:

France:Germany:Americas:

Spain:India:

China:Email:Web:

Workshopby José M. Caballero (Jose.caballero@trendcomms.com)

Triple Play tester, Triple Play testing, Triple Play installation, Triple Play maintenance, Triple Play commissioning, Triple Play tester, Triple Play testing, Carrier Class Triple Play installation, Triple Play maintenance, Triple Play tester, Carrier Class Ethernet, Metro Ethernet tester, Metro Ethernet testing, Metro Ethernet installation, Metro Ethernet maintenance, Metro Ethernet commissioning, Carrier Class Ethernet tester, Carrier Class Ethernet testing, Carrier Class Ethernet installation, Carrier Class Ethernet maintenance, Gigabit Ethernet tester, Gigabit Ethernet testing, Gigabit Eth-

ernet installation, Gigabit Ethernet maintenance, Gigabit Ethernet commissioning, Gigabit Ethernet protocols

TriplePlay Services & Protocols

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About Trend Communications

Trend Communications is an international company supplying hand-held test equipment and on-line monitoring systems to the communications market. Trend’s solutions are intended to cover businesses involved with broadband access, voice, datacom, network management, photonics transmission, metropolitan and mobile networks.

Trend has always been at the forefront of the communications test market, and our strength is based on the robustness and high quality of our products. Our solutions combine excellence and high technology with ease of use, covering such technologies as Triple Play, xDSL, 3G/UMTS, ISDN, IP, Carrier Ethernet, NG SDH/SONET and NGN.

At Trend our mission is to be the preferred supplier of Field Deployable Testers through innovative design and cost leadership.

Trend Communications is a subsidiary of IDEAL INDUSTRIES, INC.

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Triple Play Tests for Residential Subscribers

• Residential subscribers buy services rather than connectivity facilities• The requirements of the residential subscribers can be summarized by the expression “Triple Play”• Quality of Experience (QoE) tests check, not only the network, but also encoding and signalling

SDH NG

IP Network

SDH NG

IP Network

Gateway

IPTV Servers

Telephones

Proxy

IPTV TestVoIP Test

Signalling

MediaSDH NG

Data Test

Routers

Internet

IP Network

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Triple Play Testing

Triple Play is an application that runs over a large stack of telecom/datacom protocols. This means that bad quality of service, or loss of service, can be caused by many different factors.

• CPE faults• Access faults depending on the technology used• IP networks must support proper QoS and multicast requirements• Service availability and performance

SDH NGIP NetworkGateway

IPTV Servers

TelephonesProxy

Internet Internet

CPE Access ServiceIP Network

- Configurations- Wiring- Hardware/Software

- xDSL/cable/fibre faults- Bit rate expectations- Security/Privacy

- Packet loss, Delays- QoS management

- IP continuity- Service availability

- Core infrastructures - Contention- Multicast - Performance- Data Performance

- Voice/Video quality- DSLAM performance

Modem

STB

VoIP

Data

DSLAM

Cable

Data over IP

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Triple Play Data

Typical data applications (generally Internet-based) that need to be to checked are:

• Web browsing performance, a basic facility for residential customers• FTP capacity for file uploading and downloading• Traffic statistics compiled during data browsing• PPP authentication

SDH NGIP Network

Internet InternetModem

Data

Data Servers

CPE Access ServiceIP Network

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Layer 3 Continuity

In the case of service failure, the following should be tested:

• Physical layer continuity - Copper pair- Fibre optics- Coaxial cable

• DSL synchronization• IP Ping continuity• Trace Route delays

IP ping

Trace Route address, timeaddress, time address, time

SDH NGIP Network

Internet InternetModem

Data

Data Servers

CPE Access ServiceIP Network

IPTV and Video Services

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IP Video Protocols

IP

UDPTCP

RTP/RTCP

Access

Application

IP suite

Signalling IPTV (Video + Voice + Data)

IGMP

RTSP Transport Stream

MPEG-1 MPEG-2 MPEG-4

EthernetPPP

ADSL2+ Fibre 802.3ahVDSL2 FTTx WiMax

NG SDH Transport

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Video Services and MPEG

The Moving Picture Experts Group (MPEG) is a working body within the ISO that is responsible for developing video and audio encoding standards for digital television delivery, IPTV, commercial advertisements and multimedia digital video applications.

1. MPEG-1, rates up to 1.856 Mbit/s• Coding of moving pictures and associated audio for digital storage media• Used in CD Video• Video resolution, generally 352 x 240/288 • MP3 is an audio subset of MPEG-1

2. MPEG-2, rates from 2 to 9 Mbit/s • Rates generally around 4 Mbit/s with ADSL2+• Video resolution generally 720 x 480, 720 x 576 or 544 x 576• Used in Cable, DBS, DVD, VoD and HDTV• When used with HDTV, MPEG-2 typically runs at 19.3 Mbit/s

3. MPEG-4, rates from 5 kbit/s to 10 Mbit/s• Developed by the ITU to enable wireless single-user video services• Mobile/POTS 5 kbit/s to 64 kbit/s• Internet 64 kbit/s to 364 kbit/s• Broadcast/VoD 364 kbit/s to 10 Mbit/s

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MPEG2 Transport Stream

The most basic component in IPTV is known as an Elementary Stream (ES). A programme contains a combination of elementary streams (typically video, one or more for audio, control data, subtitles, etc.)

Each ES that is, output by MPEG audio, video and data encoders contains a single type of signal. There are various forms of ES, including: Digital Control Data, Digital Audio, Digital Video and Digital Data.

The ES is sent to a processor that creates a stream called a Packetized Elementary Stream (PES). This is then broken into fixed-size Transport Packets (TP) and combined with the Transport Stream (TS).

The TS used in MPEG-1, MPEG-2 and MPEG-4 is designed to allow multiplexing of digital video and audio, and also to synchronize the output. It has error correction features for transport over imperfect media. The MPEG-2 TS is the transport format, but the contents are not necessarily MPEG-2.

Distribution AudioVideo

DataAudio

MPEG-2Processors

MPEG-2 TSCompressorsNetwork

MPEG-2 TSDecoder

Triple Play TV

Multiplexer

ES PES TSTP TS TV

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MPEG2 Transport Stream

• PID=000: PAT - Programme Association Table, lists the PIDs of tables describing each programme.• PID=001: CAT - Conditional Access Table, defines the type of scrambling used + management info.• PID=X: PMT - Programme Map Table, defines the set of PIDs associated with audio, video, data...)• PID=010: NIT - Network Information Table, contains details of the bearer network used• PID=Y: PES - Packetized Elementary Stream, each independent sequence of voice, video or data

4 bytes

PayloadHeader

184 bytes

Adaption field

Transport Scrambling ControlPID

Transport PriorityPayload Unit Start Indication

Transport error indication Sync byte

Adaption Field Control Continuity Counter

PID

8 1 1 1 13 2 2 2

Transport

Transport

TP Header

AudioVideo

DataAudio

PAT table

- Network Info: PID=10- Program H: PID=306- Program X: PID=032- Program Z: PID=510

PMT table (per programme)

- Video: PID=160- Audio Spa: PID=234- Audio Fren: PID=233- Subtitle Eng: PID=237

Packet

Elementary Stream

TransportPackets

Stream

PES

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MPEG Frames

• I-Frames (Intra) can be reconstructed without any reference to other frames.• P-Frame (Prediction) only contains the differences compared to the original I-Frame. The source and

the receiver run the same algorithm, the source receives the error and creates a P frame to fix it.• B-Frames (Bidirectional prediction) contain differences for moving objects; they are fed with preceding

and following P frames (forward and backward prediction).

In theory, the number of B-frames that may occur between any two I- and P-frames is unlimited. In practice, there are typically up to twelve P- and B-frames occurring between each I-frame. One I-frame will occur approximately every 0.4 seconds during video showtime.

I-Frame

Motion

I-Frame

+

Prediction

P-Frame

Error Compensation

Creation of the P frame at source sideI

BB

P

B

B

B

PB

B

PI

forward prediction

bidirectional prediction +

time

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Multicast Services

To efficiently deliver traditional TV channels, the IP multicast method is used, where all the users that subscribe to a programme will receive the same signal. To save bandwidth, hosts must tell their nearest router, by using the Internet Group Membership Protocol (IGMP), that they want to listen to a certain group.

Router

WWWClient

WWWServer

HTTP requestHTTP answer

MailSender

ServerMailRelay

Mail

Route path

Route path

IPTVServer

ClientIPTV

ClientIPTV

UNICAST: Web browsing APPLICATION LAYER MULTICAST: E-mail

NETWORK LAYERMULTICAST IPTV

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Channel Zapping Delay

IGMP snooping is a method for intelligent forwarding of multicast packets within a layer-2 broadcast domain. IGMP registration information is snooped to create a distribution list of workstations in order to know which end-stations will receive packets with a certain multicast address.

Channel Zapping IGMP is a test used to measure the delay that occurs when a user joins or leaves a specific multicasting group. In other words, it is an IPTV channel zapping measurement.

Originator Switch

without IGMP snooping

Originator

with IGMP snooping

Multicastagent

Multicastagent

Multicastagent

Join request

Join reply

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Measuring the Subscriber’s Experience

Quality of Experience (QoE) is how good the service is from the customer’s point of view. Often the QoE of a customer is only brought to the Service Provider’s attention when the customer complains or when subscription revenues start to fall.

Testing the QoE:

• A combination of packet impairments and video content measurement• Standards based measurements. Set up parameters and PASS/FAIL results,

including: IGMP latency, TS Continuity errors, TS Interarrival Jitter and PCR Jitter.• Emulation of Video Servers/Client and IGMP

- Pixelation- Freezing- Lip Sync- Blurring- Distortion- Zapping Delay

AudioVideo

DataAudio

Contribution

IP NetworkSTB

Packet LossSync error

Packet DelayPacket Delay

Continuity error

PCR JitterTransport error

Packet Jitter

QoS parameters User Experience QoE measurement

Coding error

- Pixelation- Freezing- Lip Sync- Blurring- Distortion- Zapping Delay

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IPTV Measurements

• Quality of Experience – Calculation of the video quality perceived by the user• Programme Rate – Transport stream rate in kilobits per second as observed• Programme Clock Rate Overall Jitter of synchronization stream in microseconds• Jitter Arrival – Number of frames discarded due to jitter• Out of Sequence – Number of miss-ordered frames delivered• In Sequence – Number of correctly ordered frames delivered• Multicast Join Time – Actual time the stream was joined in “epoch” seconds• Multicast First Time – Actual time the first data arrived in “epoch” seconds• TS packets & TS total packets received counter

- TS Video packets & video packets counter- TS Audio packets & TS Audio packets counter- TS Data packets & TS Data packets counter- TS Null packets & TS null packets counter

VoIP

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IETF VoIP Protocols

• SIP is used at the control plane and RTP/UDP is used for the voice transport• H.323 was the first, and still is, the most used (but it is getting less popular),

easy internet working POTS & ISDN• SIP, used for IP phones, is currently popular as it is very flexible• SIP can be integrated easily with PCs, e-mail, web and corporate platforms

IP

UDPTCP

RTP

Audio VideoSDP PINT IMP

SIPN

etw

ork

Ap

plica

tion

Tra

nsm

ission

Supplementary

Signalling Voice over IP

Services

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VoIP Market

1. Residential VoIP• Generally a routed IP service• Usually finishes in a VoIP gateway that connects it to the PSTN

2. Enterprise VoIP• Service is set up across the VPN by adding VoIP facilities to the PBX or by upgrading routers and PCs• E1/T1, FRL or ATM Hub and Spoke architectures can be upgraded

Mapping in Frames

SDH NGPE1

PE2

PE4

PE3

Enterprise VoIP

Hub & Spoke

VoIP

VoIP

VoIP

Mapping in Frames

SDH NGPE1

PE2

PE4

PE3

Meshed

VoIP

VoIP

VoIP

Residential VoIP

VoIP PSTN

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Voice over IP Network

IP is data-oriented, but can also support multiple applications based on voice and video.

Why VoIP?

• Subscribers: cheaper calls, integration with PCs and e-mail• Carriers: convergence across a unique network• Service providers: new business opportunity• Manufactures: new market demands

VoIP uses known protocols such as:

• IP, TCP, UDP (User Datagram Protocol)• RTP (Real Time Protocol), RTCP (Real-Time Control Protocol)• SIP (Session Initiation Protocol), H.323 (ITU-T)

Hi!

IP Network

PSTN

PABXVoIP Router Router

Gateway

VPN

VoIP

Voice Codec Framing Protocols

Codec

Transcoding

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Types of Voice Codecs

Wave Form codecs

• PCM (Pulse Code Modulation) G.711, ISDN: 8b x 8k/s = 64 kbit/s, CD: 16b x 44 k/s x 2ch = 1.408 Mbit/s• ADPCM (Adaptive Differential PCM) G.726/27, 40 / 32 / 24 / 16 kbit/s• CVSD (Continuously Variable Slope Delta)

Vocoders, or synthetic voice

• LPC (Linear Predictive Coding)

Hybrid codecs, wave form with synthesizer

• CELP (Code book Excited Linear Prediction)• ACELP (Algebraic CELP)• RPE-LTP (Regular Pulse Excitation - Long-Term Prediction)• VSELP (Vector-Sum Excited Linear Prediction)

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Voice Codecs

• Wave form codecs: PCM, ADPCM, CVSD• Synthesizer or Vocoders: LPC• Hybrids: CELP, ACELP, RPE-LTP, VSELP

1 2 4 8 16 24 32 40 64

1

2

3

4

5

MOS

CS-ACELP 8(G.729)

LD-CELP 16(G.728)

ADPCM

PCMVOCODERS

CELP

ADPCM 24(G.726)

ADPCM 16(G.726)

PCM(G.711)

ADPCM 32(G.726)

LPC 4.8

kbit/s

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Voice Codecs: User Satisfaction

R-factor: in-service voice quality measurement based on observed traffic flow for every phone call

MOS: Mean Opinion Square. To arrive at an MOS score, a tester assembles a panel of “expert listeners” who rate the quality of speech samples that have been processed by the system under test.

• Ideally, a panel would consist of a mix of male and female listeners of various ages• The samples should reflect a range of typical voice conversations• The panel rates the quality of the system output from 1 to 5, with 1 indicating the worse and 5 the best• The scores of the panelists are then averaged

R factor

1009490

80

70

60

50

MOS

0

4.54.44.3

4.0

3.6

3.1

2.6

1.0

Very Satisfied

Satisfied

Some users dissatisfied

Many users dissatisfied

Nearly all users dissatisfied

Not recommended

Desirable

Acceptable

Not acceptablefor toll quality

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Out of Service Measurements

PSQM (Perceptual Speech Quality Measure), defined by ITU-T P.861, uses pre-recorded voice signals that are transmitted at the origin and compared at reception in the 300 - 3 400 Hz frequency range. Created to evaluate codec performance, basically the distortion of the speech signal. PSQM is not designed to reflect the effects of network packet loss or jitter.

PAMS (Perceptual Analysis Measurement System) also compares an output signal with the input signal, but using a different algorithm based on factors of human perception to measure voice quality, scoring on a 1 to 5 scale that can be correlated to MOS.

PESQ (Perceptual Evaluation of Speech Quality), developed based upon PAMS and an improved version of PSQM called PSQM+. Uses the best features of both: the robust time-alignment techniques of PAMS with the accurate modelling of PSQM. It targets not only VoIP, but also ISDN, GSM and POTS.

Distribution Network

PABXVoIP Router Router VoIP

Signalgeneration

Signalreception

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SIP Protocol

SIP is the protocol used to establish IP sessions between users, to set up VoIP calls, and also multimedia conferences, multimedia distributions or multicast sessions. However, SIP does not transport voice or multimedia contents.

IP

SIP Proxy

Caller

Domain A Domain B

Router Router

Recipient

sip.caribean.comSIP Proxy

sip.trendcomms.com

SIP Signalling

VoiceINVITE

SIP Request

(simplified trace)INVITE sip: pepe@trendcomms.com SIP/2.0Via: SIP/2.0/UDP mkt12.caribean.com;To: pepon <sip:pepe@trendcomms.com>From: Alice <sip:alice@caribean.com>;Contact: <sip:leila@mkt12.caribean.com>Content-Length: 142

SIP Response

(simplified trace)SIP/2.0 200 OKVia: SIP/2.0/UDP mkt12.caribean.comTo: pepon <sip:pepon@trendcomms.com>;From: leila <sip:leila@caribean.com>;CSeq: 314159 INVITEContent-Length: 131

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Typical SIP Session

SIP protocol (IETF RFC 3261).

What it does cover

• User search• Call Init, Control and Close • IP address, UDP/TCP • Changes during the session• Supplementary services

What it doesn’t cover

• Type of network to be used• Type of codecs to be used• Session details (formats,

codecs...)• How and where the proxy,

registers, redirections etc. are implemented

Note that the caller does not send the Invite message directly to the recipient, but to an SIP proxy that locates the user and starts negotiating the session parameters.

SIP Proxy

Caller Recipient

Router Router

Invite

Trying

Invite

InviteTrying

Ringing

RingingRinging OK

OK

OKACK

VoIP session

Bye

OK

sip.ideal.com SIP Proxysip.trendcomms.com

IP Network

SIP

SIP

VoIP

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Internet working with PSTN/ISDN

Done by means of gateways.

• Gateways translate the voice between the IP and the PSTN network.

• Signalling, SIP to/from SS#7, has to be translated as well.

• Translated messages (often only approximations to the original).

There are two types of gateways:

1. Media Gateways (MGs) convert data from the format required by a circuit-switched network to that required by a packet-switched network.

2. Media Gateway Controllers (MGCs) to handle all tasks related to call control and signalling

VoIP

IP Phone Proxy

Gateways

PTSN / ISDN

Invite

InviteTrying

IAM

ACM

Trying

ANM

Session Progress

Session ProgressOne-way RTP

OK

OK

ACK

ACK

communications

One-way Circuit

Two-way RTPCommunications Two-way Circuit

Legacy

Ringing tone

Hello?

Two-way Circuit

PCM

SwitchVoice

SIP ISUP

IP Network

SignallingSignalling

Voice

MG

MGC

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Converged Telephony

• Media Gateway Control Protocol (MGCP), defined in RFC 2705, handles signalling between MG and MGC, can be used to set up, maintain and terminate calls between multiple endpoints. It enables session establishment between a PSTN phone number and an IP telephone.

• MEGACO: an enhanced version of MGCP.• ISDN User Part (ISUP/SS7) defines the protocol and procedures used to set up, manage and tear

down all circuits to carry voice and data calls over the Public-Switched Telephone Network (PSTN).• Trunking MG: gateway to interconnect PSTN and IP networks• Access MG: provides VoIP connectivity to existing T1, E1 CAS, and other legacy interfaces

ISUP

PSTN

Trunking MG

ISUP

User Agent

Switch

Legacy MG

CP

or

IP

Meg

aco

MGC

MGCP/Megaco

Access MG

MGC

SIP

ISDN

PSTN

LegacyPBX

VoIPPCM

Signalling (ISUP)

Voice (PCM-VoIP)

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RTP Basics

Real Time Protocol (RTP) RFC 3550

• Used to transport voice and video signals in real time• Statistical multiplexing (data packets) produces jitter at the far end• RTP inserts a time stamp in all voice packets • Time stamps are used to ensure that all voice packets, that are delivered to the far end, maintain the

time that was originally generated.

Note that RTP does not provide QoS, but just transports timing signals.

Encoder

RTP

VoiceDecoder

Voice stream Packets ^ timestamps Packets with jitter

Retimed stream

IP

Clock Clock

RTP Voice stream

Hellooo!

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RTP Mixers and Translators

Translators and Mixers are intermediate network systems that work at RTP level.

• Translators usually change the format of the incoming RTP packets and forward them to their final destination with their SSRC identifier untouched. The operations performed by translators include downsampling of RTP data for low-bandwidth networks, changing the encoding of video or audio streams, and tunnelling of multicast data across firewalls.

• Mixers combine streams from different sources into a single outgoing RTP stream, that is, a multiparty telephone call. The mixer combines data from different sources; it cannot choose the SSRC identifier from any of them and so puts its own SSRC in the forwarded data.

Router

Translator

DSLAM

DSLAM

High-bandwidth area

Low-bandwidth area

High rate dataLow rate

data

(rate conversion)

Mixer

DSLAM DSLAM

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Real-Time Control Protocol

RTCP complements the RTP protocol with information about the QoS received: delays, loss, jitter, etc.It provides:

• persistent session information,• basic session management,• performance feedback to communication parties of intermediate probes.

Report Report

RRSR

IP

bla, bla, BLA!

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VoIP Delays

Rec. ITU-T G.114, unidirectional delay in ms:

• 0 - 150 acceptable for most applications• 150 - 400 acceptable, but degrades the QoS• > 400 unacceptable; only for voice messages or walkie-talkie gadgets

IP

Hellooo!

Encoding Buffering Ingress

Transmission

Egress Jitter buffer Decoding

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Acronyms

ACELP: Algebraic CELPADSL: Asynchronous Digital Subscriber LineADSL2+: ultra high speed ADSLARPU: Average Revenue Per UserCELP: Code book Excited Linear PredictionCPE: Customer Premises EquipmentDiffServ: Differentiated ServicesDOCSIS: Data Over Cable Service Interface

Specification DSCP: Differentiated Services Code PointDWDM: Dense WDMEVC: Ethernet Virtual Connection EVPL: Ethernet Virtual Private Line EVPLAN: Ethernet Virtual Private LANFR: Frame RelayFTTP: Fibre To The PremisesFTTC: Fibre To The Curb FTTH: Fibre To The Home FTTN: Fibre To The NetworkGFP: Generic Framing ProtocolGPRS: General Packet Radio ServiceGSM: Global System for Mobile communicationHFC: Hybrid Fibre/Coaxial network HSDPA: High Speed Downlink Packet AccessIntServ: Integrated ServicesISP: Internet Service ProviderISUP: ISDN User PartLCAS: Link Capacity Adjustment SchemeLLC: Logical Link ControlLSP: Label-Switched PathLSR: Label-Switched Router

MAC: Media Access layerMGC: Media Gateway Controllers MGCP: Media Gateway Control ProtocolMOS: Mean Opinion Square.MPLS: Multiprotocol Label SwitchingMSPP: MultiService Provisioning platformMSTP: MultiService Transport PlatformMSSP: MultiService Switching PlatformNG SDH: Next Generation SDHNGN: Next Generation NetworkOAM Operation, Administration and

Maintenance OLT: Optical Line TerminationONU: Optical Network UnitPSQM: Perceptual Speech Quality MeasurePAMS: Perceptual Analysis Measurement

SystemPESQ: Perceptual Evaluation of Speech QualityPON: Passive Optical NetworkPSTN: Public Switched Telephone NetworkPWE3: PseudoWire edge-to-edge Emulation QoS: Quality of ServiceRPE-LTP: Regular Pulse Excitation - Long-Term

PredictionRTP: Real Time ProtocolRTCP: Real Time Control ProtocolSDH: Synchronous Digital NetworkSLA: Service Level AgreementSIP: Simple Internet ProtocolSTB: Set Top BoxUTP: Unshielded Twisted Pair cableVC: Virtual Concatenation

VPN: Virtual Private NetworkVDSL: Very High Bit Rate DSLVLAN: Virtual LANVPLS: Virtual Private LAN Service VPWS: Virtual Private Wire ServiceVOD: Video On DemandVoIP: Voice over Internet ProtocolVSELP: Vector-Sum Excited Linear PredictionWiFi: Wireless Fidelity, or Wireless Local Area

Network, WLANWiMAX: World-wide Inter operability for

Microwave AccessWIS: WAN Interface SublayerADPCM: Adaptive Differential PCMCVSD: Continuously Variable Slope Delta

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That’s all, thanks