Managing VoIP Quality Issues in converged Networks -...

© 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 1

Managing VoIP Quality Issues in converged Networks

Faisal Chaudhry [email protected]

Solutions Architect Cisco Systems

Syed Khurram [email protected]

Manager, Advance Services Cisco Systems

© 2008 Cisco Systems, Inc. All rights reserved. 2

Agenda

 Architectures using VoIP  Recognizing Voice Quality Issues  Classifying Voice Quality Attributes to Root

Cause  Address Voice Quality by Implementing

Quality of Service (QoS)  Echo  Q & A


PBX PBX

Router/GW Router/GW

Call Processing Call Processing

Toll Bypass

Router/GW Router/GW

IP WAN

IP WAN

End-to-End IP Telephony with Application Enablement

IP Telephony

Enterprise ‘VoIP’ Solution Sets: Toll Bypass and IP Telephony

IP Phones IP Phones

Analog or Digital Phones

Analog or Digital Phones


Branch A IP WAN

Branch B

Centralized Call Processing

PSTN

Headquarters

Applications (VMail, IVR, ICD, ...)

CallManager Cluster

Gatekeeper GK

Applications

CallManager Cluster

Applications

CallManager Cluster

Distributed Call Processing

Single Site

Enterprise ‘VoIP’ Solution Sets: IP Telephony Deployment models


Transit/Tandem

Business & Residential

Access

Residential Access PBX

PSTN Gateway

SIP Proxy (IMS)

IP Phone

MG

Call Manager

Voice CPE

GK

H.323 Gatekeeper

Data Network

VoCable /VoETTx

Local Services Call Agent PSTN

PSTN

Service Provider Voice Architectures


SSP SLT SGW Softswitch SSP SLT SGW

Gateway

SSP SLT SGW

Gateway

Gateway

SSP SLT SGW

Gateway

Signalling : MGCP, H.323, SIP, SS7 (SIGTRAN)

VoIP Bearer traffic

Remote sites over WAN

Service Provider VoIP Transit and Class 5 Replacement


PSTN Interconnect

MxU Res. Villas

Softswitch

IP Network Triple play services

IP-IP Interconnect

SBC Session Border Controller

Service Provider Triple Play


 Managed Voice service for Enterprises by SP: On-premises or Centralized Hosted Call Control Interface to the PSTN/PLMN (Centralized or on-premises) Interface to Legacy equipment (such as PBX, Voicemail) Dial plan Management Billing Customer Premises Equipment e.g. Analog or IP Phones Need to cater for QoS, Resiliency, Security …

Service Provider Hosted and Managed IP Telephony


Customer Premises Sites

SP Hosted Infrastructure (VoIP Transit)

SS7 Signaling

TDM Voice

Voice Gateways

H.323 Gatekeepers

(or SIP) GK

LAN Switch

MGCP

GK

SP Hosted IP Telephony

Access (DSL, ETTx…)

Firewall

Media Resources (Conferencing,

Announcements etc)

Call Control

Central Voicemail

Central LDAP

Softswitch

Service Provider Hosted and Managed IP Telephony


Softswitch SS7

Signaling

TDM Voice

Voice Gateway

H.323 Gatekeepers

(or SIP Proxy) GK

LAN Switch

Firewall Access Router

Voicemail

MGCP

GK

(optional) PSTN

(optional) PSTN

Call Control

Voicemail

Call Control

Service Provider Managed IP Telephony

Customer Premises Sites

SP Hosted Infrastructure (VoIP Transit)

Access (DSL, ETTx…)


Agenda





Categorizing and Defining the Symptoms

 Noise Conversation is still Intelligible; presence of static, hum, crosstalk intermittent popping

 Voice distortion Problem that affects the voice itself Echoed voice Garbled voice Volume distortion


Noise

Clicking Cause: Clock Slips or Other Digital Errors

Crackling Cause: Poor Electrical Connection, Electrical Interference

Crosstalk Cause: Signal Leakage Due to Wires Located in Close Proximity

Absolute Silence Cause: Aggressive Voice Activity Detection (VAD)


Noise (Cont.)

Hissing Cause: VAD

Static Cause: Codec Mismatch; Enhanced by VAD


Echoed Voice

Listener Echo Cause: Long Echo Tail; Echo Canceller Is (ECAN) Not Activating

Talker Echo Cause: Long Echo Tail; ECAN Is Not Activating

Tunnel Voice Cause: Tight Echo with Some Loss


Garbled Voice

Choppy Voice Cause: Consecutive Packets Lost or Excessively Delayed Disabling DSP Predictive Insertion Where Silence Is Inserted Instead

Synthetic Voice Cause: Single Packet Loss or Delay Beyond the Bounds of the De-Jitter Buffer Playout Period

Underwater Voice Cause: A Common Cause of This Problem Is G729 IETF and Pre-IETF Codec Mismatch


Volume Distortion

Fuzzy Voice Cause: Too Much Gain on the Signal

Muffled Voice Cause: Overdriven Signal or Some Other Cause That Eliminates or Reduces Signal Level at Frequencies Inside the Key Range for Voice (Between 440 and 3500)

Soft Voice Cause: Attenuated Signal

Tinny Voice Cause: Overdriven Signal that Eliminates or Reduces Signal Level at Frequencies Outside the Key Range for Voice (Between 440Hz and 3500Hz)


Agenda



Quality of Service (QoS)  Echo  Q &A


Problem and Root Cause Association

Gain Adjustment

Packet Loss

Delay

Echo Return Loss

Jitter

VAD

Absolute Silence

Fuzzy Voice Synthetic Voice

Soft Voice Muffled Voice

Choppy Voice

Static and Hissing

Listener Echo


•  Absolute silence •  Clipping •  Static and hissing •  Underwater voice

Classifying Voice Quality Attributes to Root Cause

Quality of Service

Network Transmission

Loss Plan

VAD, Codecs

•  Loss •  Jitter •  Delay •  Synthetic voice •  Robotic voice •  Choppy voice •  Periods of silence

•  Gain adjustment •  ERL

•  Talker echo •  Listener echo •  Tunnel voice •  Fuzzy voice •  Muffled voice •  Tinny voice

Synchronization, Cabling

•  Crackling •  Clicking •  Crosstalk


Basic Guidelines for of Voice over IP  Transmit voice traffic the fastest way possible

Delay is bad (worsens echo, awkward conversations, etc.) Minimize as many sources of delay as possible Goal: keep delay to less than 150ms

 Transmit VOIP packets as a steady, smooth stream Any delay should be consistent Inconsistent delay is called “Jitter” Compensating for Jitter creates additional delay

 Drop any packets received out of order Voice does not tolerate delays…it’s better to drop the packet CODEC logic can compensate for some dropped packets

 Above all…it’s gotta sound good (subjective)


Agenda





Voice QoS Requirements End-to-End Latency

Delay Target

Avoid the “Human Ethernet”

Time (msec) 0 100 200 300 400 500 600 700 800

Hello? Hello?


Elements That Affect End-to-End Delay

IP WAN

Campus Branch Office

PSTN

Fixed (Sampling rate)

CODEC

Variable (Can Be Reduced

Using LLQ)

Queuing

Variable (Can Be Reduced

Using LFI)

Serialization Propagation and Network

20–50 ms

Jitter Buffer

End-to-End Delay (Should Be < 150 ms) As per ITU G.114 Recommendations


Voice

3

Voice

3

Voice

3

Voice

3 Reconstructed Voice Sample

Packet Loss

Voice QoS Requirements Factors which Degrade Voice Quality


  Current DSP CODEC algorithms can correct for 30 msec of lost voice by using predictor algorithm

Generally 1 G.729A voice packet contains 20 msec of voice

  Lost packets induce “clipping” and temporarily expand the jitter buffer, which increases end-to-end latency

One lost Voice over IP packet causes a MODEM retrain; Two drops can cause a call disconnect

  Causes of packet loss: Network quality, network congestion and delay variation

Voice

3

Voice

3 Voice

3

Voice

3 Reconstructed Voice Sample

Packet Loss



t Sender Transmits A C

D1 D2 = D1

Sender Receiver

PBX PBX Network

C

B

t Sink Receives

Variable Delay – Jitter

B A



 When voice call starts, the de-jitter buffer fills up to the quiescent point

As voice frames arrive too fast, the queue fills As voice frames arrive too slowly, the queue empties

 Depth of queue varies with network operation Over-/-under flow will cause gaps in speech and underwater voice

V V V V V V V V V V V V

Under Flow: Queue Empties If

Voice Frame Arrive too Slow

Fixed Playout Rate

= Codec Frame Rate

Voice Frames to DSP decode

Voice Frames From Network

Variable Arrival Rate

= Codec Frame Rate +/- Δ

Quiescent Point: Specified in mSec

V V V V V

Over Flow: Queue Fills If Voice Frame

Arrive Too Fast

Voice QoS Requirements De-Jitter Buffer Operation

Normal Operating Queue Depth


Voice QoS Requirements Summary for Voice

  Latency ≤ 150 ms   Jitter ≤ 30 ms   Loss ≤ 1%   17–106 kbps guaranteed priority

bandwidth per call   Bandwidth for Voice-Control traffic

per call SCCP: 150 bps + Layer 2 overhead

  CAC must be enabled (Integrated Services)

  Smooth   Benign   Drop sensitive   Delay sensitive   UDP priority

Voice One-Way Requirements


Enabling QoS Differentiated & Integrated Services Standards

Campus Branch Office

Central Site

SRST Router

IP WAN

PSTN

Classification: DSCP settings & attribute within a packet to differentiate (L3) RFC 3246: If mark packet with EF then PHB is to move this packet to front of queue during congestion. Mark at the edge! Trust Boundary: Define and Enforce a Trust Boundary at the Network Edge


Quality of Service Operations How Does It Work and Essential Elements

Classification and Marking

Queuing and Dropping

Post-Queuing Operations

  Classification & Marking: The first element to a QoS policy is to classify/identify the traffic that is to be treated differently. Following

classification, marking tools can set an attribute of a frame or packet to a specific value.   Policing:

Determine whether packets are conforming to administratively-defined traffic rates and take action accordingly. Such action could include marking, remarking or dropping a packet.

  Scheduling (including Queuing & Dropping): Scheduling tools determine how a frame/packet exits a device. Queuing algorithms are activated only when

a device is experiencing congestion and are deactivated when the congestion clears.

  Link Specific Mechanisms (Shaping, Fragmentation, Compression, Tx Ring) Offers network administrators tools to optimize link utilization


Classification Tools Ethernet 802.1Q Class of Service

  802.1p user priority field also called Class of Service (CoS)

  Different types of traffic are assigned different CoS values

  CoS 6 and 7 are reserved for network use

TAG 4 Bytes

Three Bits Used for CoS (802.1p User Priority)

Data FCS PT SA DA SFD Pream. Type

802.1Q/p Header

PRI VLAN ID CFI

Ethernet Frame

1

2

3

4

5

6

7

0 Best Effort Data

Bulk Data

Critical Data

Call Signaling

Video

Voice

Routing

Reserved CoS Application


Classification Tools IP Precedence and DiffServ Code Points

  IPv4: Three most significant bits of ToS byte are called IP Precedence (IPP)—other bits unused

  DiffServ: Six most significant bits of ToS byte are called DiffServ Code Point (DSCP)—remaining two bits used for flow control

  DSCP is backward-compatible with IP precedence

7 6 5 4 3 2 1 0

ID Offset TTL Proto FCS IP SA IP DA Data Len Version Length

ToS Byte

DiffServ Code Point (DSCP) IP ECN

IPv4 Packet

IP Precedence Unused Standard IPv4

DiffServ Extensions


Classification Tools DSCP Per-Hop Behaviors

  IETF RFCs have defined special keywords, called Per-Hop Behaviors, for specific DSCP markings

  EF: Expedited Forwarding (RFC3246) (DSCP 46)

  CSx: Class Selector (RFC2474) Where x corresponds to the IP Precedence value (1–7) (DSCP 8, 16, 24, 32, 40, 48, 56)

  AFxy: Assured Forwarding (RFC2597) Where x corresponds to the IP Precedence value

(only 1–4 are used for AF Classes) And y corresponds to the Drop Preference value (either 1 or 2 or 3) With the higher values denoting higher likelihood of dropping (DSCP 10/12/14, 18/20/22, 26/28/30, 34/36/38)

  BE: Best Effort or Default Marking Value (RFC2474) (DSCP 0)


IP Precedence and DSCP diffs

ToS Byte P2 P1 P0 T2 T1 T0 CU1 CU0

•  IP precedence—three bits (P2 to P0) •  Delay, Throughput and Reliability—three bits (T2 to T0) •  CU (Currently Unused)—two bits(CU1-CU0)

DiffServ Field DS5 DS4 DS3 DS2 DS1 DS0 ECN ECN

•  DSCP—six bits (DS5-DS0) •  ECN—two bits •  The standardized DiffServ field of the packet is marked with a value so that the packet receives

particular forwarding treatment or PHB, at each network node.


Example of IP precedence and DSCP IP Prec 5 (101) maps to IP DSCP 101 000 ToS Byte

1 0 1 T2 T1 T0 CU2 CU0 DiffServ Field

1 0 1 0 0 0 ECN ECN

Precedence Level Description 7 Stays the same (link layer and routing protocol keep

alive) 6 Stays the same (used for IP routing protocols) 5 Express Forwarding (EF) 4 Class 4 3 Class 3 2 Class 2 1 Class 1 0 Best effort


Assured Forwarding DSCP chart

Drop Class 1 Class 2 Class 3 Class 4 Low 001010

AF11 DSCP 10

010010 AF21 DSCP 18

011010 AF31 DSCP 26

100010 AF41 DSCP 34

Medium 001100 AF12 DSCP 12

010100 AF 22 DSCP 20

011100 AF32 DSCP 28

100100 AF42 DSCP 36

High 001110 AF13 DSCP 14

010110 AF23 DSCP 22

011110 AF33 DSCP 30

100110 AF43 DSCP 38


WAN

QoS is Needed to Minimize Packet Loss, Delay and Delay Variation

Where QoS Is Needed Central Site Remote Branch

•  Layer 3 Policing •  Multiple Queues on All Ports; Priority Queuing for VoIP •  WRED Within Data Queue for Congestion Management

•  Define trust boundary •  Classification on IP Phone and Access Switch •  Multiple Queues on IP Phone and Access Ports

•  Classification and Trust Boundaries on IP Phone , Access Layer Switch and Router •  Multiple Queues on IP Phone and All Access Ports

QoS—Campus Access QoS—Campus Distribution

•  Low-latency Queuing •  Link Fragmentation and Interleave •  Bandwidth Provisioning •  Admission Control

QoS—WAN QoS—Branch


Queuing on the IP Phone (Edge device)

COS = 5

COS = 0

COS = 5

COS = 7

Untrusted: Phone ASIC Will Re

-Write CoS = 0

Mini-switch inside the IP Phone re-tags the packets from access port In your network, what edge device is used? L2 or L3? Type of L2?


•  Priority Queue (PQ) is for VoIP RTP traffic (CoS = 5)

•  Round-Robin Scheduling with a PQ Timer

TX Voice RTP Packets

Queue 0 (PQ)

Queue 1 Queue 2 Queue 3

Queue ? RX

Queuing on the Edge device (IP Phone)


QoS in Campus


Is QoS Needed in the Campus?

“Just throw more bandwidth at it. That will solve the problem!”

Maybe, Maybe Not; Campus Congestion Is a Buffer Management Issue


Campus QoS Considerations Typical Campus Oversubscription Ratios

Access

Distribution

Core


Si Si

Area’s Where QoS Maybe a Concern

TX TX TX TX TX TX Access

Distribution

Core

Access, Distribution and Core Queuing

  Output buffers can reach 100% in campus networks   When an output buffer congests, dropped packets occur at the

ingress interfaces   QoS required when there is a possibility of congestion in buffers   Multiple queues are the best way to “guarantee” voice quality


Tx Buffer Congestion

Ethernet Switch

Voice

To Core Data

Data

Data RX

RX

RX

RX

TX

Data Flows “Hog” Tx Buffer

Additional Flows, Including Voice,

Can Not Get Access to Tx

Buffer


Guaranteed Voice Requires Multiple Queues

Ethernet Switch

Voice

To Core Data

Data

Data RX

RX

RX

RX

Queue Assignment Based on CoS/ToS

Classification

Voice Put Into “Delay and Drop” Sensitive Queue

TX


Core

Distribution

Access LAN Switch

LAN Switches

MGCP Gateway

H.323 Gateway

Call Control (e.g. Softswitch)

4/2

Campus QoS—Access Layer

 Call Signaling is control plane for VoIP traffic

 All VoIP Control Plane Traffic should be Classified as DSCP=AF31 in the VoIP Gateway or from the LAN Switches e.g.

SIP: UDP 5060 Skinny Control: TCP 2000-2002 MGCP Control: UDP 2427 H.323 Control: TCP 1720 TCP

11000- 11999

Control and Management Plane Traffic

4/3 4/4


QoS in WAN


WAN Edge QoS Design Considerations QoS Requirements of WAN Aggregators

WAN Aggregator (WAG)

WAN Edges

Campus Distribution/Core

Switches

LAN Edges

WAN

Queuing/Dropping/ Shaping/Link-Efficiency Policies

for Campus-to-Branch Traffic


Scheduling Tools Queuing Algorithms

  Congestion can occur at any point in the network where there are speed mismatches

So no reordering if no congestion!

  Scheduling tools define how packet exits a device Only activated when congestion occurs

  Cisco IOS-based software queuing in WAN routers Low-Latency Queuing (LLQ) used for highest-priority traffic (voice/video) Class-Based Weighted-Fair Queuing (CBWFQ) used for guaranteeing

bandwidth to data applications

Voice

Video

Data 3 3

2 2

1 1


3 1 2 3 0 2 0 2 1 2 0 1

TAIL DROP

3

3

3

WRED

0 1

0

1

0

3

Queue

Scheduling Tools Congestion Avoidance Algorithms

  Queueing algorithms manage the front of the queue i.e. which packets get transmitted first

  Congestion avoidance algorithms, like Weighted-Random Early-Detect (WRED), manage the tail of the queue

i.e. which packets get dropped first when queuing buffers fill   WRED can operate in a DiffServ compliant mode which will

drop packets according to their DSCP markings


Link-Specific Tools Link-Fragmentation and Interleaving

  Serialization delay is the finite amount of time required to put frames on a wire

  For links ≤ 768 kbps serialization delay is a major factor affecting latency and jitter

  For such slow links, large data packets need to be fragmented and interleaved with smaller, more urgent voice packets

Serialization Can Cause

Excessive Delay

With Fragmentation and Interleaving Serialization Delay Is Minimized


Voice

CBWFQ Fragment

Interleave

WFQ

Link Fragmentation and Interleave

Low Latency Queuing

Packets Out

Packets In

Police

Video PQ

Best Effort

Signaling

TX Ring

Critical Data

Layer 3 Queuing Subsystem Layer 2 Queuing Subsystem

PQ Packets Do Not Go Through Fragmentation: on Low Link Speeds,

You Cannot Put Large (i.e., Video) Packets in the PQ with Voice Packets

Scheduling Tools LLQ and Class_Based-WFQ Algorithm


Enabling QoS in the WAN Traffic Shaping

Line Rate

R

Traffic Shaping Limits the Transmit Rate to a Value (R) Lower than Line Rate

Without Traffic Shaping

With Traffic Shaping

Why Is It Needed 1.  Line Speed Mismatch 2.  Remote to Central Site

Over-Subscription 3.  To Prevent Bursting Above

Committed Rate (CIR)

1

2

3

Central Site

Frame Relay or ATM

E1

E1 E1 64 kbps E1

CIR = 64 kbps Remote Sites

3 2 1


Link Capacity = (Min BW for Voice + Min BW for Video + Min BW for Data) / 0.75

Voice Is Not Free—Especially on Low Speed Links—Engineer the Network for Data, Voice, and Video

Sources of Trouble for VoIP

Sum of Traffic = 75%

Data Voice Routing

etc

Link Capacity

Video Voice/Video Control

LLQ = 33%

Provisioning


Call Admission Control (CAC)


Introduction Why Is Call Admission Control (CAC) Needed?

PSTN

Circuit-Switched Networks

Packet-Switched Networks

PBX

Physical Trunks

STOP

IP WAN Link

IP WAN Link’s LLQ Is Provisioned for Two Calls (Equivalent to Two “Virtual” Trunks)

Third Call Rejected

No Physical Limitation on IP Links; Third Call Can Go Through, but Voice Quality of All Calls Degrades Call Admission Control Blocks Third Call

IP WAN

Router/ Gateway

Call Control


“site” N “site” 2 “site” 1

Topology-Unaware CAC

 Vendor specific implementations

static configuration in call processing agent

 Define the logical “site” to match physical site

 Configure a max number of calls or max amount of bandwidth

...

Headquarters

Branch 1 Branch N Branch 2

Call Processing

Agent

Max. 5 Calls

Max. 3 Calls

Max. 8 Calls


Topology-Unaware CAC

 Simple hub-and-spoke topologies  Simple MPLS-based topologies

Hub Site

Spoke Site 1

Spoke Site 2

Spoke Site N ...

Spoke Site 1

Spoke Site 2

Spoke Site N

Spoke Site Y

Spoke Site X

or SP-Provided MPLS IP WAN

Limited to:


Topology-Unaware CAC “Real” Network Topology Aspects

Spoke Spoke

3rd Tier Hub

Spoke Spoke Spoke Spoke

3rd Tier Hub

2nd Tier Hub 2nd Tier

Hub

Spoke Spoke

Spoke Spoke Spoke Spoke

3rd Tier Hub

3rd Tier Hub

2nd Tier Hub 2nd Tier

Hub

Spoke Spoke Spoke Spoke Spoke Spoke

2nd Tier Hub

2nd Tier Hub

Dual-Homed Branch Sites

Redundant Hub Sites

Dual WAN Links

Fully Meshed Core

Hub-to-Hub Links

Multi-Tier Architectures

1st Tier Hub 1st Tier

Hub

1st Tier Hub

Core Core Multiple Paths


Topology-Aware CAC Integrated Services

  Based on communication between call processing agents and the network on available resources

  Can be applied to any WAN network topology

  Dynamically adjusts to topology changes

  Requires a signaling protocol

  RSVP (Resource ReSerVation Protocol) is the first industry standard for QoS signaling

IP WAN (Any Topology)

RSVP

Call Processing

Agent

Call Processing

Agent


Call Admission Control Principles Summary

  Based on static configuration within call processing agent

  Does not react to network topology changes (e.g. link failures)

  Limited to simple topologies   Limited to a single call

processing agent   Examples: Cisco

CallManager® “static” locations, Cisco IOS®

gatekeeper zones

  Based on real network resources

  Requires a signaling protocol (RSVP)

  Reacts to network topology changes

  No topology limitation   Can be used by different call

processing agents   Examples: Cisco

CallManager RSVP-enabled locations, Cisco H.323 & SIP Voice gateways (AS5x00, 3800 etc), Cisco multiservice IP-IP gateway

Topology-Unaware CAC Topology-Aware CAC


48 24 0

24

30 6

0

64

64 40

40

RSVP Principles

96

56

24 72 80

72 48

30 24

24 64 48 48

24

48

64 24

80 RSVP Bandwidth Pool Provisioned on Each

Router Interface with: ip rsvp bandwidth ...

If There Is Sufficient Bandwidth Throughout

the Network, the Reservation Succeeds

If Bandwidth on Any Link Throughout the Network Is

Not Sufficient, the Reservation Fails

Device 1

Device 2

RSVP Signaling Uses Same IP Route as the

Data Stream That Needs Reservation

Legend: = kbps

Remaining in RSVP Bandwidth Pool

56

RSVP-Unaware Routers Ignore and Forward All

RSVP Messages

Device 3

Device 4


End devices CAC in H.323 & SIP

PSTN Cisco ASAP PSTN PSTN

GK

Gateway Resources: PSTN Side CAC

D S 0

DSPs

HDLC framers

D S 0

DSPs

HDLC framers

• Calls not delivered/accepted if any set of required resources are all in use

• Resources monitored – DS0s, DSPs, HDLC framers

• Hairpin, Play announcement or – Release with ‘Resource Unavailable’


PSTN Cisco ASAP PSTN PSTN

GK

H.323 Gateway Resources: Network Side CAC

RAI

1 RAI - Resource Availability Indicator

D S 0

DSPs

0%

100% High Low

DS0

, DSP

re

sour

ces

• H.323: Gateway informs Gatekeeper if resource threshold exceeded using RAI1 –  GK selects GW based on RAI status

• ICPIC2 or Delay & Loss thresholds

2 ICPIF - Calculated Planning Impairment Factor (ITU G.113)

End devices CAC in H.323 & SIP


General Guidelines for QoS in the WAN

  Use Low Latency Queuing (LLQ) anytime VoIP over the WAN is involved

  Traffic shaping is a requirement for frame-relay/ATM environments

  Use Fragmentation & Interleaving (LFI) techniques for all links below 768Kbps

Don’t use LFI for any video over IP applications   Properly provision the WAN bandwidth   Call admission control is a requirement where VoIP calls can

over-subscribe the provisioned BW   Use cRTP carefully


Agenda





Talker Echo

  Talker echo occurs when a talker's speech energy, transmitted down the primary signal path, is coupled into the receive path from the far end; the talker then hears his/her own voice, delayed by the total echo path delay time; if the ‘echoed’ signal has sufficient amplitude and delay, the result can be annoying to the customer and interfere with the normal speech process; talker echo is usually a direct result of the 2-wire to 4-wire conversion that takes place through ‘hybrid’ transformers

John’s Voice

Echo of John’s Voice

Talker Echo (Most Common)

John San Jose

Jane New York


Listener Echo

  Listener echo occurs at the far-end by circulating voice energy; again, listener echo is generally caused by the 2W/4W ‘hybrid’ transformers; caused by the “echo being echoed”; the talker’s voice is echoed by the far end hybrid and when the echo comes back to the listener, the hybrid on the listener’s side echoes the echo back towards the listener; the effect is the person listening hears the talker and an echo of the talker

John’s Voice

Delayed Echo of John’s Voice

Listener Echo (Less Common)

John San Jose

Jane New York


Sources of Echo

+-

Tx

Rx

2-Wire Subscriber Loop

Xfmr

4-Wire Trunk Rs Rs ZL

Z?

Schematic of a Hybrid Circuit

Any Mismatch of Impedance vs. ZL Will Cause the Tx Signal to Appear

on Rx, This Is Hybrid Echo

Hybrid Echo

Earpiece

Microphone

Echo

Acoustic Echo


What Makes Echo a Problem?

 An analog leakage path between analog Tx and Rx paths

 Sufficient delay in echo return for echo to be perceived as annoying

 Sufficient echo amplitude to be perceived as annoying

For Echo to Be a Problem, all of the Following Conditions must Exist:


Tail Circuit

Ecan Terminology

IP Network

Rin Rout

Sin

H(t)

y(t)

-

Echo Return Loss (ERL) is the echo level loss provided by the tail circuit. That is, if an speech signal entering the tail circuit from the network at a level of X dBm0 (i.e. Rout – above), the level of the echo coming back from the tail circuit into the Sin terminal of the echo canceller is (Rout – ERL) dBm0 i.e.

ERL

ERL = Rout - Sin


Ecan Terminology Tail Circuit

IP Network

Rin Rout

Sin

H(t)

Sout

y(t)

-

ERLE Echo Return Loss Enhancement (ERLE) refers to the additional echo loss obtained through the operation of the echo canceller. An echo canceller is not a perfect device, and the best it can do is attenuate the level of the returning echo. ERLE is a measure of this echo attenuation through the echo canceller.

ERL

It is the difference is level (in dB) between the signal arriving from the tail circuit at the Sin terminal of the echo canceller and the level of the signal leaving the echo canceller (and entering the network) at the Sout terminal.

ERLE = Sin - Sout


Ecan Terminology Tail Circuit

IP Network

Rin Rout

Sin Sout

H(t)

y(t)

-

ACOM

ERLE

ACOM (aka Acombined) is simply the total echo return loss seen across the Rin and Sout terminals of the echo canceller. It is the echo return loss seen by the network.

ERL

Rin

Sout

ACOM = ERL + ERLE


 Give the echo canceller enough information to distinguish between echo and normal conversation; the only parameters you have control over are:

Input level (input gain) Output level (output attenuation) Echo canceller coverage

E C A N

Tail Circuit

IP Network

Rin Rout

Sin Sout -

ACOM

ERLE

ERL

Output Attenuation

Input Gain

Eliminating Echo


Rules of Thumb

 Echo on one end is typically generated at other end

 Bits don’t leak Echo is not introduced on digital links Introduced by 2 to 4 wire conversion in hybrid and impedance mismatch or via acoustic feedback

 ERL must be 6dB for ECANs to engage  Be careful setting echo-cancel coverage;

Longer coverage yields longer convergence time; Configure the coverage so that it is long enough to cover the worst-case for your environment, but no higher


Q and A


References

  Echo

http://www.cisco.com/univercd/cc/td/doc/cisintwk/intsolns/voipsol/ea_isd.htm#91601

  Voice Quality Degradation Symptoms http://www.cisco.com/warp/public/788/voice-qos/symptoms.html#clip

  Quality of Service http://www.cisco.com/warp/public/732/Tech/qos/

  IP SLA

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_book09186a00802b2a6c.html

  VoIP Troubleshooting Using “show call active voice” http://www.cisco.com/warp/public/788/voip/show_call_act_voice.html

Managing VoIP Quality Issues in converged Networks -...

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