VOIP Troubleshooting using ClearSight Analyzer™

36
VOIP Troubleshooting VOIP Troubleshooting using ClearSight using ClearSight Analyzer™ Analyzer™

Transcript of VOIP Troubleshooting using ClearSight Analyzer™

VOIP Troubleshooting VOIP Troubleshooting using ClearSight using ClearSight

Analyzer™Analyzer™

SLIDE 2

Agenda

ClearSight – Company Overview

ClearSight Analyzer Overview

VoIP Analysis with ClearSight Analyzer

Signaling Protocols

Media Protocols

Hands on Examples

Q & A

SLIDE 3

ClearSight – Company Overview

Founded March, 2001 as AppDancerFounders created “Sniffer Pro”

Name/Product Launch – October, 2003HQ: San Mateo, CaliforniaFiscal 2004:

• 6 Figure Operating Income, on• Mid-7 Figure Revenues

Worldwide Partnership - Spirent CommunicationsReceived Key Editorial Accolades

ClearSight Analyzer™ClearSight Analyzer™ OverviewOverview

SLIDE 5

Traditional Network Analyzers:

Use The “Bottom-Up Approach”

Troubleshooting Based Upon Raw Packets – Not Necessarily Real-Time

Focus on lower-layer elements –Physical Errors

Host Tables

Top Talkers

Utilization

Neither Intuitive Nor Easy!Neither Intuitive Nor Easy!

SLIDE 6

Traditional Analyzer GUI

SLIDE 7

The “ClearSight” View

SLIDE 8

Bridging The Gap

ClearSight’s Analyzer Enhances the IT Professional’s Ability To:

Quickly Identify Trends, Issues and Problems Determine if They’re•Physical •Environmental •User - Related

Facilitates Deployment of Optimum Resource Levels

SLIDE 9

Special Edition

Product Portfolio

ClearSight Analyzer

1 Gigabit

10 Gigabit

NEM Operator Enterprise

Software-Only

Full Duplex

Segment Appeal

Hard

ware

P

erfo

rman

ce

Distributed

SMB

VoIP TechnologyVoIP Technology

SLIDE 11

VoIP  ( Voice over IP)PBX  ( Private Branch Exchange)GW ( Gate Way)PSTN ( Public Switched Telephone Network)

PBX

IP packet

Fixed-line telephone

IP phone

PBX

PBX/GW

PSTN

IP networ

kIP phone

PBX/GW

( VoIP)

( The conventional telephone )

VoIP Technology

Fixed-line telephone

Fixed-line telephone

Fixed-line telephone

Difficult to Troubleshoot

SLIDE 12

VoIP Signaling Protocols

The protocols which manage setup, transfer, maintenance, and disconnect of a call in an IP telephoneProtocols which exchange information on terminal capabilities and/or CODEC’sTypical VoIP signaling protocols

SIPH.323Cisco SkinnyMGCPMEGACO

SLIDE 13

Physical Layer

Data Link Layer

Network Layer ( IP )

UDP

RTP

RTCP

MovieH.26x

VoiceG.7xx

UDP ( TCP )

SIP   (Session Initiation

Protocol)

SDP (Session Description Protocol)

Signaling protocolsMedia protocols

VoIP Protocol Stack (SIP example)

SLIDE 14

ClearSight Creates a Call List

ClearSight creates a group of associated protocol sessions for each call –

Also automatically creates a call list

SLIDE 15

SIP Request Method & Response

<Method>INVITE : Used to initiate the sessionACK :   Used as a confirmation that the final response has been receivedBYE :   Used to terminate a sessionREGISTER : Used by a user-agent client to log in and register its address with a SIP server     OPTIONS : Queries a server as to its capabilitiesCancel : Used to terminate a pending<Response>1xx Provisional2xx Success3xx Redirection4xx Request Failure5xx Server Failure6xx Global Failure

SLIDE 16

SIP Ladder Display

< Direct communication between two terminals >

SLIDE 17

SIP Server Example

SIP server172.17.0.20

IP phone 100172.17.1.15

IP phone 100172.17.1.15

SIP SIP

RTP

<Communication through a SIP server>

SLIDE 18

SIP Ladder Display

< Communication through a SIP server>

SLIDE 19

Voice ( A <- B )

RTP

RTCP

Voice ( B -> A )

RTP

RTCP

Video( A <- B )

RTP

RTCP

Video ( B -> A )RTP

RTCP

AB

●RTP ( real time transport protocol ) RFC1889RTP is designed to provide end-to-end network transport functions for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services.●RTCP ( RTP control protocol ) RFC1889RTCP is companion protocol of RTP. This is used for quality feedback of the voice packets.

Media Protocols : RTP & RTCP

SLIDE 20

RTP Flow Statistics

SLIDE 21

Major Elements of Voice Quality

Packet loss< 1% :  Negligible impact3-5% : Momentary pauses in conversations10% :  Considerable deterioration of quality> 20% :  Difficult to carry on any

conversation

Latency> 25ms : Requires echo cancellation< 150ms : Typically no discernable problem150-400ms : Although the voice quality of user application is affect

ed, use is possible if you understand it.

> 400 ms : Unsuitable for most applications.

JitterEcho

SLIDE 22

The subjective evaluation method

MOS (Mean Opinion Score) ITU-T recommendation P.800

The objective evaluation method using Perception modelPSQM (Perceptual Speech Quality Measurement) ITU-T recommendation

PAMS (Perceptual Analysis Measurement System) British telecom

PESQ (Perceptual Evaluation of Speech Quality) ITU-T recommendation P.862

The objective evaluation method using Calculation modelR-Value (E-Model)   ITU-T recommendation G.107

PhoneIntermediate device Phone

IP Network

It Talks It Talks

IP Network

Test Equipment

Voice File

Deteriorated Voice SignalOriginal Voice Signal

Voice Quality Evaluation Method

Intermediate device

Intermediate device

Intermediate device

SLIDE 23

Scoring R-Value and MOS

VoIP AnalysisVoIP Analysiswithwith

ClearSight Analyzer™ClearSight Analyzer™

SLIDE 25

ClearSight VoIP Advantages

Multiple Physical Segment Correlation (Real-time and post analysis)

The next generation trouble-shooting tool for data and voice convergence

Real-time VoIP playback – including distributed remote

The only trouble-shooting tool supporting video playback

Extensive playback CODEC support

Real time QOS, VoIP alarms and triggers

Integrated VoIP SLA

Reporting: VoIP Summary, SLA Trending, and Per-conversation QOS Report

VoIP Application level filter – Eg.; phone number

VoIP/WiFi QOS analysis over AP and WiFi LAN switch

SLIDE 26

VoIP QoS Report

SLIDE 27

VoIP Playback

ClearSight Supported CODECs for playbackAudio CODECsG711(U-law / A-law), G.729, G.723, G.722, etc.

Video CODECsJPEG(411, 422, 111), H.263 Mode A/B

SLIDE 28

VoIP Reporter

ClearSight Supported CODECs for playback

Audio CODECsG711(U-law / A-law), G.729, G.723, G.722, etc.

Video CODECsJPEG(411, 422, 111), H.263 Mode A/B

ClearSight Analyzer™ClearSight Analyzer™Hands on ExamplesHands on Examples

SLIDE 30

SIP Phones can ring each other, however, Phone A→B: No problemPhone B→A : I can’t hear you !

Q. WHY ?                  

OK !

OK!

Ring!!

Ring!!

Phone A

Phone B

Phone B Phone A

I can’t hear anything !I can hear you.

NG !

OK !

Solving a Typical VoIP Problem

SLIDE 31

ClearSight

Phone A〔 10.0.0.10〕

SIP Server192.168.0.3

Phone B〔 10.0.2.10 〕

L3 SwitchL3 Switch

Router Router

Cloud

CHub

VoIP System

SLIDE 32

Signaling Flow

SLIDE 33

WLAN Switch- AirSpace- ArubaEtc.

HUB

SIP Server

ClearSight

AP1AP2

move

WLAN SIP phone 1

Initial RTP flowRTP flow after roaming

ClearSight solution for VoIP over WLAN Switch

WLAN SIP phone 2 WLAN SIP phone 2

-Voice quality analysys

- VoIP signaling analysis

- Roaming analysis

WLAN packets over Ethernet encapsulated with special

header

SLIDE 34

AireSpace Decode Sample – packets between AP and WLAN Switch

LWAPP Encapsulation header

SLIDE 35

VoIP ladder display of WLAN VoIP phone traffic – traffic between AP and WLAN Switch

The Next Generation The Next Generation VoIP Trouble-shooting VoIP Trouble-shooting

ToolTool