VoIP Technology OverviewVoIP Technology Overview

Introduction to VoIP Technology & Testing

OutlineVoIP basicsMarket overviewVoIP technology basics

Control plane – call setupData plane – the conversation

Testing & Deploying VoIP

The Basics

What is VoIP?Traditional PSTN voice calls use dedicated TDM circuits over SONET/SDH networks. These networks are optimized for voice traffic.VoIP = Voice over Internet Protocol (layer 3)

Calls are broken down into IP packets using CODECs, and transmitted over networks as data (not voice).Packets are re-assembled at the receiving end and the CODEC returns the call to analog form.

Initiating Receiving

CODEC

Jitterbuffer

CODEC

Packets from a single conversation may take many different routes and arrive at irregular intervals. They are buffered before being converted back to analog speech.

InternetInternet

The Basics

Where is VoIP?VoIP calls can be placed on any device which can run a CODEC:

PC’s (using a ‘softphone’ - Skype, Yahoo Messenger, Vonage, PC-based PBX)A dedicated IP-phoneCell phones: digital cell phones already transmit packetized voice– and are moving to more popular VoIP CODECs to support wireless roaming to WiFi networks.

The Basics

What else is VoIP?VoIP has cool features… once a session is started, you can:

transfer files, pictures, or videos to the other partystart a video conferencing sessionadd others to your call (dynamic teleconference)start an instant messaging session

Voicemail / videomail in your emailVoIP numbers follow you; any IP connection can be your phone – long distance means ?

Great, where can I get some?Almost anyone with an IP network is offering VoIP to homes and business clients:ISPs ● cable MSOs ● telcos…

The VoIP Explosion

†† Infonetics Aug. 05

VoIP Test Equipment Market

35% Growth through 2010MSOs: 50M VoIP lines by end-2005Enterprises: a 3.5B market!VoIP available from most Telcos/MSOs

Drivers:xDSL in 2005: ↑ 45% to 13.8M lines (FCC)Broadband access to businesses ↑ 34%Carrier Ethernet to grow 400% by 2009 ††

50% of access lines ● 83% of WANs

Frost & Sullivan 2004

VoIP Technology

Making a phone call means

Calling someone → call setup or session initiation→ also known as signaling→ uses the network control plane (layer 4)→ key protocols: SIP, H.323→ requires a signaling gateway / proxy server

Having a conversation → real-time data streaming: layer 3 packets→ uses UDP (packets not resent if lost)→ key protocol: RTP→ common CODECs: G.711, 723-6-8-9→ requires a media gateway

1

2

VoIP Sessions

Signaling and streaming use different network paths and resources

Image courtesy of the SIP Forum

Call Setup

Signaling processEstablishes a connection between 2 or more participantsOnce session is registered, real-time data streaming beginsSession is managed and closed by the signaling protocol

Typical Call Setup & Session Control

Described visually by ‘Ladder-Diagrams’

Call SetupSignaling protocols

SIP is becoming the standard for audio callsH.323 will remain big in video conferencingMGCP / NCS is commonly used for voice over cable applicationsOthers: Skinny (per Cisco), MGCP / Megaco & SCTP (for TDM calls over IP)

Session Initiation Protocol – managed by the SIP Forum / IETFWorks similar to HTTP + TCP / IP ● Still developing as a standardFor voice, video conferencing, media streaming, data sharingIdentifiers can be a phone number, IP or email address

10 year old protocol for video conferencing (and VoIP)Used for years on ISDN-based video conferencing systemsMature – in its 5th major revisionComplex – SIP is much simpler and equally capable

NCS = Network Call Signaling – the CableLabs (PacketCable) version of MGCPMGCP = Media Gateway Control Protocol, an IETF IP-telephony signaling protocol designed for IP phones that are lower cost than SIP or H.323 phones.MGCP/MEGACO uses softswitches for call control and more resembles circuit-switched PSTN than do SIP and H.323.

SIP

H.323

MGCP/ NCS

Audio-Streaming

RTPReal-time Transport Protocolpackets containing encoded voice traffic transmitted over UDP

Variations & ExtensionscRTP – compressed RTP– point-point onlyRTCP – Real-Time Control Protocol

Data plane protocol transmitted next to RTP traffic. RTCP lets phones exchange QoS info to adjust call-quality in real-timeWorks with VoIP analyzers – allows quality testing using any phone as an endpointRTCP is not often enabled for phone calls, since it requires extra bandwidth

RTCP XRA newer version of RTCP: more efficient transmission of standard QoS metrics: round-trip delay, packet loss, echo, quality

Audio-Streaming

CODECsEnCOder / DECoder – A/D and D/AThere are many kinds of CODECs in the world:

For images: jpeg and GIFFor file compression: ZIP filesFor video MPEG-2, 4For audio MP3, wma, wav

There are standard VoIP CODECs (e.g. G.711), & proprietary ones (common in cellular networks)Type & bitrate used depends on bandwidth (space) available & the quality required… like MP3 filesThe CODEC used is specified in the RTP packet header…but is transparent to transmission… only the phones need to know the CODECs.

Common CODECs

CODEC choice is usually based on available bandwidth – the uncompressed G.711 CODEC can replicate toll-quality voice; G.729 uses the most compression and cannot achieve MOS > 3.8.

Data Rate = 4 KbpsLine Rate = 50 fps

IP packet size = 50 bytesCompression on voice channel to:- Voice rate = 8 Kbps- VoIP payload = 10 bytes

VoIP G.729

Data Rate = 6.4 KbpsLine Rate = 33.3 fps

IP packet size = 64 bytesCompression on voice channel to: - Voice rate = 6.4 Kbps- VoIP payload = 24 bytes

VoIP G.723.1

Data Rate = 64 KbpsLine Rate = 80 fps

IP packet size = 140 bytesNo compression algorithm on the voice channel- Voice rate = 64 Kbps- VoIP payload = 100 bytes

VoIP G.711

ValuesDescriptionPre-defined Type-P packet

VoIP Service Quality

Quality of Service (QoS)VoIP is prone to quality issues: packet-based protocols were not designed to deliver real-time dataNetworks can be optimized for VoIP, but most are notFor good QoS, VoIP traffic needs:

to traverse a network with solid packet transmission performance

high-priority in routing and switching queues (low latency)enough bandwidth for the number of simultaneous calls expected

a network with low packet-jitterFinely tuned VoIP gateways, echo cancellers, Analog Telephone Adapters (ATAs), etc.

OC-48

Gig-E Switch

Switch

DWDM

Router

Customer A

Customer C

Customer C

Customer A

Customer B

Customer B

Problems are usually caused by problems on backhaul & access links, by Customer Premises Equipment (CPE), and softswitch / media gateway configuration.

50

60

70

80

90

100

2.5 3 3.5 4 4.5

MOSr

R

Common QoS Metrics

MOSMean Opinion ScoreCan also be determined using various methods using IP-based or analog measurements, or a combination of both.Scale from 1 → 55 is theoretical best. 4.2 is a good PSTN call. Cell phones usually have MOS of ~3.5

Commonly used indices represent the audio quality of a call. These scores do not help troubleshoot or optimize a network – they only indicate if things are working well or not. A variety of detailed measurements are required to fully assess the quality of voice services and the underlying service degradation causes.

R FactorPractical range from 50-95From the ITU-ratified E-ModelConsiders more impairments than most MOS algorithmsThe most accurate E-model implementations use both analog & IP measurements to calculate metricsE-Model also returns MOSr, and several other indices.

R-factor and the E-model derived MOS are related

MOS / QoS Algorithms

PSQMPerceptual Speech Quality MeasurementDesigned for CODEC testing, does not account for network issues

PESQPerceptual Evaluation of Speech Quality, based on an ITU standardPESQ gives MOS and other call quality metricsImprovement on PSQM, accounts for distortion, echo, etc.Partially accounts for network issues, but not excessive jitter or packet loss.

VQESAn alternate voice quality algorithm developed by MCI labs gives VQES MOS and percentage of unsatisfied users, P(UDI) – an estimate of a callers’ experience over the complete call duration.

MOS algorithms are based on psycho-acoustic research, and are derived from subjective speech quality assessment studies conducted on a sample of people listening to audio samples. Each algorithm has advantages and applications they are best suited for.

Testing VoIP

Lab ApplicationsRequires high load-generation capability (traffic simulation) and signaling simulation – often on specialized hardware.Should be complemented by a true end-end voice quality test system (analog-analog ports) to account for CODEC / echo canceller / ATA effects.

Provisioning VoIPA variety of handheld portable solutions are available. The most effective communicate with a centralized test management server to log results and standardize pass/fail criteria.

VoIP Monitoring SystemsActive test systems place test calls through the network and accurately assess end-end service quality. Automation schedules tests for continuous monitoring. Active testing is also used for provisioning and troubleshooting applications.Passive monitoring systems ‘sniff’ network traffic and assess VoIP quality from IP-based metrics. Complements active testing, but cannot fully account for quality on mixed PSTN-IP networks, or when data is encrypted.

Testing VoIP quality is critical, given that people expect toll-quality voice from any subscription-based phone service. Research, provisioning and monitoring applications require different approaches.

Active Test LayoutsTesting can be conducted using different test architectures. Each has advantages in certain applications. The most sophisticated test automation platforms support multiple layouts - using a variety of techniques offers the best test coverage.

Advanced On-Net QoS Testing and Monitoring

2 ProbePP PP

ResponderPP R

Day-of-Install & Troubleshooting Tests in the Field

Single-EndPP

Off-Net Testing -No Far-End Probe Required

LoopbackPP

Test to Subscriber MTAs & CATV Transponders

InteractivePP

Interactive Test Agents for Customer Support

OEM TesterOEMPP

Test to Handheld Test Sets & Network Elements

or or…

or

or

or

Test ProbeTest Server

Conclusions

VoIP is in full deployment globallyCustomers are becoming more knowledgeable and demand quality voice service.As networks are not optimized for real-time data, testing is critical at all stages of deployment:

Day of installContinuous monitoringTrouble-ticket resolution

The most effective test methods use multiple test layouts and centralize data management from provisioning, monitoring, and troubleshooting.

To try a Minacom test system live, visit: www.minacom.com/demoweb

Which service would you like to test today?

info@minacom.comwww.minacom.com

Head Office

260 Queen StreetMontreal QC H3C 2N8CANADA

Tel.: +1 514 879 9111Fax: +1 514 879 9333

Customer Service Locations

• United States• United Kingdom• France• Japan• Korea• Italy• Sweden• Cyprus• Brazil

• Singapore• Taiwan• Russia• Argentina• India• Estonia• Qatar• Pakistan

Information subject to change without notice. Minacom, Simplifying Service Assurance, DirectQuality and PowerProbe are registered trademarks of Minacom. All other trademarks used in this document are the property of their respective owners.

© 2005 Minacom. All rights reserved.

Scott SumnerDirector of Marketing

Voice Fax Video Internet VoIP