Transcript VoIP QOS and QOE

Pre-deployment Engineering for Voice
over IP Solutions
(IPT implementation in NUI Galway)
Pat Dempsey
Head of Strategic Services
NUI Galway
Email: [email protected]
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
HEAnet Workshop
November 2006
Voice Over IP Is a Unique Application Demands Intelligent Handling
PERFORMANCE DIMENSIONS
APPLICATION
Sensitivity to
Bandwidth
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
1These
applications are highly loss sensitive but loss is managed by TCP retransmissions
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November 2006
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
Design Guidelines / Traffic
Engineering – next 4 slides
We will focus on these for
the rest of presentation
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November 2006
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
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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
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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
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Is customer Network Ready for
VOIP - Perform a Network
Assessment
> Health Check – NUIG
used NetIQ
> Pinpoint misconfigurations prior to
deploying a single
phone
> Can WAN links
support G.711 or
G.729?
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November 2006
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
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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
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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
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Measuring QoE: MOS and the EModel
> Mean Opinion Score
(ITU P.800)
•
Subjective call quality measurement
perceived by the user
R-Value User Satisfaction MOS
5.0
100
94
90
Very Satisfied
4.4
4.3
Satisfied
> 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
80
Toll Quality
Some Users
Dissatisfied
4.0
3.6
70
Many Users
Dissatisfied
3.1
60
Nearly All Users
Dissatisfied
2.6
50
0
Not
Recommended
1.0
Average quality scores over the duration ofAdapted
a call
may not reflect end users
from Diagram by Roger Britt, Senior Eng., Nortel
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perception of call quality
November 2006
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
Workshop
Some QoS technologies are end-to-endHEAnet
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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
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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.
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Real Time Protocol
RTP 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)
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RTCP XR
IETF RFC 3611 - Focus on End User
Experience
> 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
IP Phone
IP Phone
CODEC
RTCP
XR
IP Network
RTCP
XR
CODEC
RTCP XR: Real-Time Control Protocol eXtended Reports
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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.
Where it all fits in!
Components of
IP Telephony Systems
Call Server
Meridian
CS1000M1
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.
Signaling Server
Media Gateway
The signaling server was introduced to
provide the IP intelligence to register,
manage, and direct IP components.
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.
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Migrating an Existing Location to support IP
Existing Meridian
Option X1 PBX
Migrated
CS1000M
Signaling Server
+ Signaling Server +
New
Software
(Release 4.5)
=
IP Enabled
CS 1000M
Supporting
IP and
all previous
services
Migrate all previous
features/services to
support analog,
digital and IP.
Administration
(Digital)
Courtesy
(Analog)
Administration Courtesy
(Analog)
(Digital)
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Executive
(IP)
Flexible Telephony Deployment in NUI Galway:
> We have choice: TDM, Hybrid IP with new
multimedia applications)
IP
Analog/Digital
Phones
Signaling
Server(s)
CS1000M
Analog/Digital
Phones
Signaling
Server(s)
=central dialplan
IP phone services
IP Phones
(up to 15,000)
Media
Gateway
PSTN
CS 1000E
Call Servers
LAN
IP &
Digital
LAN
IP
Phones
(up to 15,000)
Digital
WAN
CallPilot,
Branch Media Gateways
Meridian 1
Digital
Phones
Analog
Phones
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