Pre-deployment Engineering for Voice over IP Solutions (IPT implementation in NUI Galway) Pat Dempsey Head of Strategic Services NUI Galway Email: [email protected]

Pre-deployment Engineering for Voice over IP Solutions

(IPT implementation in NUI Galway)

Pat Dempsey

Head of Strategic Services

NUI Galway

Email: [email protected]

http://www.nuigalway.ie/vp/siea/internal/documents/nuigalway_logo_colour_300.zip

HEAnet Workshop November 2006

Objectives of this session

> Understand how IP network design can impact the quality and reliability of VoIP services

> Understand the basic factors and design concepts for designing the IP network to support VoIP traffic

> Calculate typical bandwidth requirements on the users IP WAN based on voice services requirements

> What Tools are available to configure/troubleshoot



1These applications are highly loss sensitive but loss is managed by TCP retransmissions

Voice Over IP Is a Unique Application - Demands Intelligent Handling

APPLICATION

PERFORMANCE DIMENSIONS

BandwidthSensitivity to

Delay Jitter Loss

IP Telephony Low High High Med

Video Conferencing High High High Med

Streaming media Low-High Med Low Med

Client / Server Transactions Low Med Low High1

Email (store/forward) Low Low Low High1

Best Effort Traffic Low-Med Low Low Low



Delivering Quality of Experience

> A satisfactory level of perceived voice quality is achieved through the following:• a properly-engineered network• good network equipment and redundancy• adequate bandwidth for peak usage• use of QoS mechanisms• ongoing monitoring and maintenance

We will focus on these for the rest of presentation

Design Guidelines / Traffic Engineering – next 4 slides



Design Recommendations for VoIP - typical

> The following slides are typical considerations when designing for VOIP

• Vendors only supports customers with Layer 2/3 switched networks (no shared media devices, cable-based, hub-based LAN)

• L2 switch ports must be set to autonegotiate for VoIP devices• Goal of Zero Percent Packet Loss for VoIP

• Use G.711 CODEC when possible• Excellent Voice Quality• Bandwidth usually available in LAN and MAN

• Use G.729A or G.729AB to conserve bandwidth• Take care to meet customer voice quality requirements• Watch out for multiple transcodings (multiple VoIP hops)• Be careful with VAD – subject to clipping effects• Centralised voice mail and music can be a call quality issue



Traffic engineering process - typical

> For site pairs, determine voice “trunks” needed

> Calculate VoIP bandwidth demands• Traffic Bandwidth Calculator / Vivinet Assessor

> Overlay VoIP traffic patterns onto physical network diagram• Vivinet Assessor

> Size the required primary and alternate converged network links:• Evaluate current traffic demand• Calculate, add in VoIP traffic demand• Evaluate various failure scenarios• Factor in desired headroom, unusable bandwidth



Bandwidth Example

> Requirement: A company wants to support up to 4 simultaneous voice calls over the IP WAN network (128kbps) between two sites

> If all 4 calls were simultaneously active, this would require 108.8 kbps (using a G.729 codec, 20 ms voice sample, and PPP overhead/frame) of the available 90 kbps of the 128 kbps link

> This requirement exceeds the carrying capacity of the link and completely starves that data traffic

> The solution is to upgrade the WAN connection bandwidth. A 256 kbps link is the minimum speed to provide 109 kbps for four G.729 VoIP calls, 80 kbps for data, and 20% availability for zero-bit stuffing



Is customer Network Ready for VOIP - Perform a Network Assessment

> Health Check – NUIG used NetIQ

> Pinpoint mis-configurations prior to deploying a single phone

> Can WAN links support G.711 or G.729?



What hurts VoIP Call Quality?

> Multiple transcodings of compressed voice • Tandem hops, voice mail compression

> End-to-end delay • Budget 250ms for G.711 • Budget 150ms for compression CODECs (G.729)

> Jitter – variable arrival interval between packets• Late packets = Lost packets

> Packet Loss• Our network likes to throw things away rather than forward

damaged goods• Overloaded queue situations, device just can’t hang onto packet

> Goal: Design Network and PBX to minimise the effects of the parameters above



IP/Packet Networks – Why QoS?

> IP networks do not guarantee that bandwidth will be available for voice calls unless QoS mechanisms are used• QoS to restrict delay, minimize packet loss

> QoS techniques can be applied to support VoIP with acceptable, consistent and predictable voice quality

> QoS mechanisms refer to packet tagging mechanisms and network architecture decisions on the TCP/IP network to expedite packet forwarding and delivery



QoS versus QoE

• Quality of Experience (QoE) is subjective and relates to the actual perceived quality of a service by the user• This applies to voice, multimedia, and data

• Quality of service (QoS) is an optimization tool designed to deliver a certain Quality of Experience (QoE) by ensuring that network elements apply consistent treatment to traffic flows as they traverse the network



Measuring QoE: MOS and the E-Model

> Mean Opinion Score (ITU P.800)

• Subjective call quality measurement perceived by the user

> E-Model (ITU G.107)• Transmission planning tool for

estimating user satisfaction• Objective measurement• E-model output: R value

• Under 60 is not acceptable• Over 94.5 is unattainable in

VOIP

R-Value User Satisfaction MOS

Not Recommended

Nearly All Users Dissatisfied

Many Users Dissatisfied

Some Users Dissatisfied

Satisfied

Very Satisfied

0

50

60

70

80

9094

100

1.0

2.6

3.1

3.6

4.0

4.3

4.4

5.0

Toll Quality

Adapted from Diagram by Roger Britt, Senior Eng., Nortel Average quality scores over the duration of a call may not reflect end users perception of call quality



What are the Choices for QoS?

There are several ways to deliver QoS, including the following:

> Network QoS Technologies• Ethernet 802.1Q/802.1p• IP Differentiated Services (DiffServ)• ATM CoS • PPP Fragmentation and Multi-Class Extensions• MPLS for Traffic-Engineered Paths

> VoIP Application QoS Technologies• Codec Selection• VAD / Silence Suppression• Call Admission Control / Bandwidth Management• Packetization rate• Jitter buffer size

Some QoS technologies are end-to-endSome QoS technologies are end-to-end



QoS Management: Ongoing Monitoring

> Passive Monitoring• Source code integrated into endpoints (i.e. Telchemy Agent in Phone)• Software performs real time, in-call quality calculation • Metrics can be obtained at end of call or mid call• Alerts in real time for voice quality degradation

> Active monitoring• NetIQ performance endpoints generate synthetic voice traffic• Useful for ongoing assessment of network and troubleshooting



Phone Diagnostic Capabilities

> Ping and Traceroute• The administrator can execute the Ping or Traceroute command from a

specific endpoint with any arbitrary destination, typically another endpoint or Signaling Server.

> IP Networking statistics• The administrator can view information on the packets sent, packets received,

broadcast packets received, multicast packets received, incoming packets discarded, and outgoing packets discarded.

> Ethernet statistics• The administrator can view ethernet statistics (for example, number of

collisions, VLAN ID, speed and duplex) for the IP Phone on a particular endpoint. The exact statistics will depend on what is available from the IP Phone for the specific endpoint.

> UNISTIM statistics• The administrator can view RUDP statistics (for example, number of

messages sent, received, retries, resets, and uptime) for the IP Phones.

> Real time Transport Protocol statistics• The administrator can view RTP/RTCP QoS metrics (for example, packet loss,

jitter, etc.) while a call is in progress.



Real Time ProtocolRTP and RTCP

> Real-time transport protocol (RTP)• Provides end-to-end delivery for voice and video on top of UDP• Maintains packet sequence

> Real-time transport control protocol (RTCP)• Specified in same IETF standard, RFC 1889• Monitors and controls information of the RTP session (not an independent

protocol)• Separates flow - RTP port number +1• Transmits packets as a percentage of session bandwidth (min. of every 5

seconds)



> Passive voice quality monitoring notifies network managers of quality degradation in real-time, expediting problem resolution

> Proactive thresholds identify problems before they are perceptible to the user and impact end-user productivity

> Granular statistics supply accurate metrics for troubleshooting and SLA delivery• Jitter, latency, packet loss, jitter buffer discards• Accurate MOS and R-value

CODEC

IP Phone

CODEC

IP Phone

RTCP

XRIP NetworkRTCP

XR

RTCP XRIETF RFC 3611 - Focus on End User Experience

RTCP XR: Real-Time Control Protocol eXtended Reports



Where it all fits in!

Meridian 1 Components ofIP Telephony Systems

Media Gateway’s have always been a part of the core TDM PBX. Formally referred to as IPE shelves in a Meridian 1, Digital Cards/ Analog Cards and Trunks reside here.

Media Gateway

Call Server

The Call Server has been in existence since the inception of the PBX. Acting as the “brains” of the PBX, it provides all of the core telephony features and functionality.

Signaling Server

The signaling server was introduced to provide the IP intelligence to register, manage, and direct IP components.

CS1000M

Once the IP Components have been added and the software is upgraded to Rel. 3 or higher, the system is referred to as a Communication Server 1000M or CS1000M.



Migrating an Existing Location to support IP

Existing Meridian Option X1 PBX

+ +

Administration(Digital) Courtesy

(Analog)

New Software

(Release 4.5)Signaling Server

MigratedCS1000M

=

Administration(Digital)

Migrate all previous features/services to support analog, digital and IP.

IP EnabledCS 1000M

SupportingIP and

all previous services

Courtesy(Analog)

Signaling Server

Executive(IP)



Flexible Telephony Deployment in NUI Galway:

> We have choice: TDM, Hybrid IP with new multimedia applications)

WAN

CS 1000ECall Servers

SignalingServer(s)

MediaGateway

IPPhones

(up to 15,000)

Analog/DigitalPhones

IP Phones(up to 15,000)

AnalogPhones

IP

IP & Digital

CallPilot,

DigitalMeridian 1

SignalingServer(s)=central dialplanIP phone services

CS1000M

PSTN

Digital Phones

Analog/Digital Phones

Branch Media Gateways

LAN

LAN


Documents

Pre-deployment Engineering for Voice over IP Solutions (IPT implementation in NUI Galway) Pat Dempsey Head of Strategic Services NUI Galway Email: [email protected]