EZQoS-3200 VOIP Monitor Administrative Guide Administrative...Getting Start 7 2 Getting Start After...

EZQoS-3200 VOIP Monitor AdministrativeGuide

Release 2.0 Aug., 2021

About EzVoiceTek

Ezvoicetek Co., Ltd. was founded by a team of specialist in the area of CTI,Contact Center, IVR, VOIP and Telecommunications. We concentrate toprovide the IPV6+IPV4 SIP server farm solution including SIP proxy server,IP-PBX, SIP surveillance server and Qos Monitor to our partner, systemintegrator and value added reseller. All Ezvoicetek solutions are provided tosupport both IPV4 and IPV6 dual stack simultaneously. We provides apainless migration path from IPV4 to IPV6 network . EzVoiceTek Co., Ltd.

1F., No.16, Ln. 257, Yuantong Rd., Zhonghe Dist., New Taipei City 235, Taiwan (R.O.C.)WEB: www.ezvoicetek .comEMAIL: sales@ezvoicetek .com

Technical SupportEmail: support@ezvoicetek .com

All rights reserved. No parts of this work may be reproduced in any form or by any means - graphic,electronic, or mechanical, including photocopying, recording, taping, or information storage and retrievalsystems - without the written permission of Ezvoicetek.

Products that are referred to in this document may be either trademarks and/or registered trademarks ofthe respective owners. Ezvoicetek make no claim to these trademarks.

EzVoiceTek Co., Ltd. (EzVoiceTek) makes no representations or warranties with respect to the contentshereof. In addition, information contained herein are subject to change without notice. Every precautionhas been taken in the preparation of this manual. Nevertheless, EzVoiceTek assumes no responsibilityfor errors or omissions or any damages resulting from the use of the information contained in thispublication..

Copyright Notice and Disclaimer

4

Table of Contents

Part I Introduction 6

................................................................................................................................... 61 Overview

Part II Getting Start 7

................................................................................................................................... 71 Logon the system

................................................................................................................................... 72 Change Default Password

................................................................................................................................... 83 Setting Sniffing Parameter

................................................................................................................................... 104 Enable Alerting Service

................................................................................................................................... 115 Create Alerting Criteria

Part III Using the System 12

................................................................................................................................... 121 Home

................................................................................................................................... 122 System

......................................................................................................................................................... 13Sniffer

......................................................................................................................................................... 18Filter

......................................................................................................................................................... 19Quality Thresholds

......................................................................................................................................................... 21Customer

......................................................................................................................................................... 23Provider

......................................................................................................................................................... 25Database

......................................................................................................................................................... 26License

......................................................................................................................................................... 26Web Service

.................................................................................................................................................. 28SSL Certif icate

.................................................................................................................................................. 29Customize Logo

.................................................................................................................................................. 29Web Login Blocked IP

......................................................................................................................................................... 30Voice Logging

................................................................................................................................... 313 Monitor

......................................................................................................................................................... 32Call Traffic

......................................................................................................................................................... 32Call Quality Split

......................................................................................................................................................... 33Call Quality Trend

......................................................................................................................................................... 34Call Status Split

......................................................................................................................................................... 34Average Success Rate

......................................................................................................................................................... 35Average Call Duration

......................................................................................................................................................... 36Call Distribution

......................................................................................................................................................... 37Codec Split

......................................................................................................................................................... 38Problem Calls

......................................................................................................................................................... 39Post Dial Delay

......................................................................................................................................................... 41Video Codec Split

................................................................................................................................... 414 Calls

......................................................................................................................................................... 43Search Criteria

......................................................................................................................................................... 45Calls Detail

................................................................................................................................... 495 Register

......................................................................................................................................................... 50Search Criteria

......................................................................................................................................................... 51Register Detail

................................................................................................................................... 526 Alert

5

......................................................................................................................................................... 52System Alert

......................................................................................................................................................... 54Alert Criteria

.................................................................................................................................................. 54Low ASR Alert

.................................................................................................................................................. 55Low Voice Quality

.................................................................................................................................................. 56High Abnormal Calls

.................................................................................................................................................. 57Low Call Duration

.................................................................................................................................................. 58High Delay Packets

.................................................................................................................................................. 58Screened Calls

.................................................................................................................................................. 59High Packet Loss Rate

.................................................................................................................................................. 60Potential Calling Attack

.................................................................................................................................................. 61Potential Register Attack

.................................................................................................................................................. 62One-Way Voice Calls

.................................................................................................................................................. 62Long Post Dial Delay

.................................................................................................................................................. 63Register Expired

.................................................................................................................................................. 64Blocked UA Calls

.................................................................................................................................................. 65Loop Call Alert

......................................................................................................................................................... 66Screened Number

......................................................................................................................................................... 67Screened User Agent

......................................................................................................................................................... 67Report

................................................................................................................................... 687 Report

......................................................................................................................................................... 69Quality Summary Report

......................................................................................................................................................... 71Quality Performance Report

......................................................................................................................................................... 73Quality Call Split Report

................................................................................................................................... 758 Administration

......................................................................................................................................................... 75Service Control

......................................................................................................................................................... 76Clean-Up

......................................................................................................................................................... 76Account

......................................................................................................................................................... 78Web Provisioning

......................................................................................................................................................... 79Upgrade System

......................................................................................................................................................... 79Downgrade System

................................................................................................................................... 809 Commit

Part IV Appendix 81

................................................................................................................................... 811 List of Used Network Ports

6

1 Introduction

1.1 Overview

This administration guide is dedicated to SIP Call Voice Quality Monitor which is anon-intrusive sip call quality monitor and alerting system based on ITU-T G.107 Emodel and support dual IPv4/IPV6 SIP calls. For product detail spec anddescription, please refer to the company web for the latest datasheet.

Getting Start

7

2 Getting Start

After successfully installed the SIP Call Voice Quality Monitor, first of all is to login tothe web management interface. You can either using IPv4 or IPv6 address toaccess GUI management interface by using popular browser such as InternetExplorer or Firefox. Welcome to join the Qos Monitor world and enjoy it.

2.1 Logon the system

After connect the Ethernet cable into the server machine, administrator need to usea computer which had Firefox or IE installed and network connected in order toconnect to system GUI. For convenience, configuration computer is recommendedto have same subnet as server.

Start the browse, and type http://xxx.xxx.xxx.xxx:9320 or https://xxx.xxx.xxx.xxx:9321to login the web manage where xxx.xxx.xxx.xxx is the IP address of server.

After connected, you should able to see th following login page. Input the default userID "admin" and password "admin" and the validation code (CAPTCHA) to logon thesystem.

2.2 Change Default Password

The default password of "admin" is madden for easy to remember. To secure thesystem access, it is recommended to change the default password as the follows.

http://xxx.xxx.xxx.xxx:9320

https://xxx.xxx.xxx.xxx:9321

8

Click ADMINISTRATION -> Account -> admin and the following screen willappear. Input the new password at the Password and Confirm Password fields andclick the Apply button to take effective. Click logout to quit the system UI and reloginby new password for confirmation.

2.3 Setting Sniffing Parameter

After the installation, firstly need setup the sniffing parameters in order to captureyour SIP calls including SIP and RTP traffic. Make sure you have setup thecorresponding port mirror on you switch for SIP and RTP traffic or use a hub (notrecommended). Click SYSTEM -> Sniffer to change the database settings asfollows:

Click the Interface you have licensed and following will appear:

Getting Start

9

Input the following parameters: indicates restart-sniffer is required

Parameter Name Value

Interface Name eth0

SI Monitor Port 8088 5060 5062

Monitor SIP Register If you have license for register sniffer, you can set to"Monitor All".

Click Start Sniffer to start the monitor for call and register.

10

2.4 Enable Alerting Service

In order to use call quality notice feature, you need to star the alerting service first. Click ALERT -> System Alert to enable the syslog for email notice as follows:

ParameterName

Description

System AlertThreshold

The filter level to send the alert out. The default is level of"Notice".

Alert to Syslog Whether to send the system alert to syslogd server or not.

Syslog Receiver IP The syslogd server to receive the system alert.

Alert to Email Whether to send the system alert to the listed email accountor not.

SMTP Server The SMTP server for sending the system alert mail notice.

Email From The email sending account (FROM)

Email To List The email addresses to receive the system alert email.

Email Subject The email subject for the system alert notice. The variable "$HOSTIP$", Host IP address, could be used in the subjectto make the subject easy to be read (e.g. System AlertNotice from $HOSTIP$).

Email User ID The email sending account ID

Email UserPassword

The email sending account password

Getting Start

11

After input the correct information, click Apply to save and take it effective.

2.5 Create Alerting Criteria

To create an alerting notice, firstly, administrator need create a alerting criteria.When the alerting criteria was matched, the corresponding alert will be sent thoughtthe syslog or email depending on the your setting. Click ALERT -> Alert Criteriaand the following screen will appear.

Click New to add a new criteria and the following will appear:

Please refer to Alert Criteria Parameters to set the corresponding alerting criteria.And click Apply. Then the alerting notice will be monitored in 5 minutes interval.

12

3 Using the System

The administrator can logon the web GUI interface to manage the system service.The default login URL for administrator login is http://xxx.xxx.xxx.xxx:9320 or https://xxx.xxx.xxx.xxx:9321 and default login id is "admin" and password is "admin".

3.1 Home

The home page of SIP call quality monitor server will display the followingparameters.

Parameter Name Description

Product Name The product name

System Version The current running system release

Web Version The current running web release

3.2 System

The system parameters including the Sniffer, Quality Thresholds, Database, Licenseand Web Service. Click the SYSTEM and will see the setting in the left panel asfollows.

http://xxx.xxx.xxx.xxx:9320



Using the System

13

3.2.1 Sniffer

The Sniffer is the main configuration for capturing SIP/RTP traffic. The system cansupport up-to 4 network interface based on license granted. Here you need toconfigure the network interface to be captured and SIP service port should behandled. Make sure you have setup the corresponding port mirror on you switch forSIP and RTP traffic or use a hub (not recommended). Click SYSTEM -> Sniffer toview and change the settings.

Click the interface you would like to setup and the following will appear:

14

The detail of each parameter is described as below: indicates restart-sniffer isrequired


Interface Name The network interface to be used for capturing SIP/RTP traffic. It is not available for call center license.

IP Address The current interface's IPv4 and IPV6 address foridentification

UDP SIP Port The source or destination SIP service UDP portshould be considered as a SIP call.

TCP SIP Capture Whether enable SIP TCP capture or not

Using the System

15

TCP SIP Port The source or destination SIP service TCP portshould be considered as a SIP call.

Save Call Capture Whether to save SIP and RTP captured data for callsor not?Disable: no RTP/SIP will be saved.Based on Filter: Only those telephone number, IP V4/V6 address's RTP or SIP will be saved.Save All: both RTP/SIP will be saved/Save SIP Capture Only: Only SIP signaling will besaved.

Encrypting RTP Data Whether encrypt RTP data for voice or not? When itis enable, the RTP will be encrypted and more CPUpower is required.

Record UnconnectedEarly Media

It is only available when voice logging license isgranted. When it is turned on, the voice logging willinclude early media even for unconnected call.

Capture SIP Proxy Whether it is captured under the SIP proxy side ornot. If it is enabled, only RTP belongs to first leg willbe save and calculate. It is recommended to bedisabled for most of applications. Please only beturned on by FAE's recommendations.

Monitor SIP Register Whether to monitor SIP Register status and messageor not? It is only valid if you have register monitorlicense purchased. When you select based on filter,the system will only capture those data matched thefiltered target.

Register Capture SavingTime

How long the register file will be saved into DB anddisk in minutes? The default is 2 hours (120)depending on the frequency you need SIP endpoint toregister.

Max SIP Time Out (secs) The max time to wait SIP message for a call. If thereis no any SIP messages was received within thisperiod, the call will be cleaned and saved. Thedefault is 300 seconds.

Max RTP Time Out(secs)

The max time to wait RTP message for a call, if RTPis captured. If there is no any RTP messages wasreceived within this period, the call will be cleanedand saved. The default is 120 seconds.

16

Max CDR Storing Thread How many thread will be used for CDR storing. Thedefault is 4.

Debug Trace Level The debug level for tracking purpose. Please onlychange it under the supervisor of systemadministrator.

Promise Mode Whether capture all traffic (Yes) on the Ethernet oronly host it self (No). Only available for stand-alongversion.

Capture Buffer (MB) Internal buffer will be used (only change underrecommendation)

Call/Register Wait afterFinal Response

Whether wait for a while after a call is received a finalresponse for call or register (only change underrecommendation)

Register OK KeepingTime (secs)

How long the system will keep the registerinformation after SIP register is success (receive 200OK)?

Register No ResponseTimeout (secs)

How long the system will keep the registerinformation cache if the SIP register is not success(not receipt 200 OK)?

Register Variance Time(secs)

Any variance will be added to calculate whether thisregister is expired or not.

Register Cleanup Interval(secs)

The interval the system will clean up the register table.

Call Complete KeepingTime (secs)

How long the system will keep the call in cache aftercall is connected?

Call No ResponseTimeout (secs)

How long the system will keep the call in cache aftercall is not connected (such as not response)

Save Video Packets Whether save video RTP into capture or not?

TCP SIP Capture Whether enable SIP TCP capture or not

TCP SIP Port The source or destination SIP service TCP portshould be considered as a SIP call.

Capture Filter The special filter will be used for this sniffer. Pleasecontact technical support if need to use it. It is onlyavailable for stand-alone license.

Using the System

17

Register LicenseReached Threshold (%)

The threshold to send alert when current register useris over this value. It is used for know when the currentregister is reaching the license register.

Call License ReachedThreshold (%)

The threshold to send alert when current call attemptis over this value. It is used for know when the currentcalls is reaching the license calls.

Interface Status The sniffing status of this licensed interface.

SIP Proxy Setting The setting regarding to SIP proxy which is not usedfor stand-alone license.

Proxy NAT Public IPAddress

If the server is setting behind a NAT server and willservice Internet customer over NAT server such asAWS, GCE, Azure. This is the IP address for NATserver or public IP address.

Proxy IP WAN IPAddress

The local IP address for WAN interface for Proxyserver or IPPBX.

Proxy IP LAN IP Address The local IP address for LAN interface for Proxyserver or IPPBX.

Proxy RTP Start Port It is internal used only and contact technical supportfor changing.

Proxy RTP Stop Port It is internal used only and contact technical supportfor changing.

Click Start Sniffer to start the monitor for this interface.Click Graceful Stop Sniffer to save the current captured data and stop the monitor.The action could lead longer time to stop compare to Force Stop Sniffer.Click Force Stop Sniffer to stop the monitor immediately without saving thecaptured data.Click Graceful Restart Sniffer to save the current captured data and restart themonitor again.Click Force Restart Sniffer to restart the monitor without saving the captured data.

18

3.2.2 Filter

Filter will be only used when you set the capture interface for "Save Call Capture" or"Monitor SIP Register" to "Based on Filter". It is useful to analyze those informationyou need only instead of all calls or registers.

When you set the Save Call Capture to "Based on Filter", only those target(telephony number, IP address) matched the filter list's capture file will be saved. Theothers will only do the QoS calculation.When you set the Monitor SIP Register, only those target matched filter list can besaved or monitored.

Click SYSTEM -> Filter -> Telephone Number and the following screen will appear:

Click New or Modify to manage the telephone number filter as below:

Using the System

19

The filter target will be telephone number to be matched. You can select whether tosave the call, save SIP or register capture or not based on your requirements.

To manage the IPV4 or IPV6 filter target, please do the same as telephone numberfilter but input the IPV4 or IPV6 address in the target field.

3.2.3 Quality Thresholds

The Quality Thresholds is used to set the threshold to identifying the voice qualitybased on MOS value. It will also be used for reporting to tell whether call quality isgood or not. The MOS value is from 0 to 5. The following is the default settings whichmight meet most of applications. Click SYSTEM -> Quality Thresholds to viewand change the settings.

20

The detail of each parameter is described as below:


Jitter Buffer Simulation The default jitter buffer simulation used for display callquality. It can be changed on the displayed result.

Very Satisfied The call quality is very good and very satisfied by allusers.

Satisfied The call quality is good and satisfied by all users.

Some UsersDissatisfied

The call quality is OK and satisfied by most of users.

Many Users Dissatisfied The call quality is not good and many users are notsatisfied.

Nearly all usersDissatisfied

The call quality is very bad and nearly all users are notsatisfied.

Low Voice Quality MOSThreshold

Below this MOS value will be considered as a lowvoice quality call for the quick path to "Low VoiceQuality Calls" in CALLS. The default value is 2.8.

Minimum RTP Send forMOS Calculation

Only those calls have more RTP packets sent than thisvalue will be considered as a validated call and it willbe calculated into MOS statistic. The small quantity ofRTP is not good to do the MOS prediction. Thedefault value is 50.

Using the System

21


Minimum Talk Time forMOS Calculation

Only those calls have duration exceed this value will betake into the calculation of MOS statistic. Smallduration is not good to be the source of MOS statistic.The default value is 2.

PDD Threshold(Seconds)

The threshold of post dial delay will be used forreporting to decide whether the post dial delay is toolong or not. The default is 5 seconds.

3.2.4 Customer

The customer is used for reporting purpose. You can generate your quality reportbased on each customer. The customer owned telephone number, IPV4 or IPV6address will be used to identify whether the record is belongs to this customer or not. Click SYSTEM -> Customer to manage the customer settings and the following willappear:

Each customer will have its name and associated telephony number, IPV4 or IPV6address. Click All Detail to add all types of customer owned target. Click TelephonyNumber to manage the customer owned telephony number. Click IPV4 or IPV6address to manage the customer owned IP address. The following is the example ofall detail.

22

You can select the different Data Type and its associated value. Such as IPv4address and input the 10.0.01 in the Value field.

And the following is the example of IPV4 address editing for the customer.

Just simply input the IP V4 address such as 192.10.0.1 and click Apply to save it.

Using the System

23

3.2.5 Provider

The provider is used for reporting purpose. You can generate your quality reportbased on each provider. The provider owned telephone number, IPV4 or IPV6address will be used to identify whether the record is belongs to this provider or not. Click SYSTEM -> Customer to manage the provider settings and the following willappear:

Each provider will have its name and associated telephony number, IPV4 or IPV6address. Click All Detail to add all types of provider owned target. Click TelephonyNumber to manage the provider owned telephony number. Click IPV4 or IPV6address to manage the provider owned IP address. The following is the example ofall detail.

24

You can select the different Data Type and its associated value. Such as IPv4address and input the 10.0.01 in the Value field.

And the following is the example of Telephone number editing for the provider.

Just simply input the provider's telephone number and click Apply to save it.

Using the System

25

3.2.6 Database

This is for system database settings. Click SYSTEM -> Database to view andchange the settings.



MYSQL DB Server MYSQL database server IP address. The default value is127.0.0.1

MYSQL Port MYSQL database connection port. The default port is3306.

MYSQL User ID MYSQL access user ID

MYSQL Password MYSQL access password

MYSQL DatabaseName

MYSQL Database Name

Max Call DataKeeping Days

How long the call data will be kept in local storage. Thesystem will clean those data automatically.

Max Register HistoryKeeping Days

How long the register data will be kept in local storage.The system will clean those data automatically.

Log Data KeepingDays

How long the log data will be kept in the system. Over thedays, the system will automatically clean it.

26

3.2.7 License

This the license granted. It can only be used on this dedicate machine. We have noresponsibility for error, omissions or any damages resulting from the wrong use ofthe license. Click SYSTEM -> License to view or import/export the license.



Product Name The licensed product name

Serial ID The serial ID generated

Machine ID The machine ID used for the license

Feature The feature list of this license

License Key The license key generated

Expired Date The expired date for the license

3.2.8 Web Service

This page come with web GUI service settings. Click SYSTEM -> WEB Service toview and change the settings.

Using the System

27



HTTP Service Port The TCP service port for web GUI management. Thedefault value for administrator and supervisor login is9000. The default value for extension login is 80.

HTTPS Service Port The TCP service port for HTTPS (SSL) web GUImanagement. The default value for administrator andsupervisor login is 9001. The default value for extensionSSL login is 443.

Display Data Rowsper Page

Number of data rows will be displayed per page. Thedefault is 15.

Force Redirect HTTPTo HTTPS

When access to HTTP web, it will automatically redirectto HTTPS service.

Show Help Whether show HELP on the administration menu or not?

Show Hint Whether show mouse over hint or not?

Use Validation Codeon Login

Whether the system use the Validation Code when userlogin. The default value is "Yes".

Write Access LogCount

Number of authenticate failure access will write a log intoWeb provisioning report.

28


Block Access IP Count Number of authenticate failure access will block this IPaddress.

Block Access IP Time(minutes)

How long the IP will be blocked?

3.2.8.1 SSL Certificate

This is the place to configure secure HTTPS used SSL certificate. You might need toget the certificate from a supplier. Click SSL Certificate and the following willappear:

Paste the purchased SSL certificate into the Certificate Data column and you canuse view Certificate to check whether it is a correct certificate or not.Paste the purchased SSL bundle certificate CA data into Bundle CA column.Paste the purchased SSL key into Private Key column and Click Save to makesystem need HTTPS SSL.

Click "Apply To Whole System" to set all SSL certificate for whole installed systemwithin the chassis. Please note every service will be restated to make the SSLcertificate effective.

Using the System

29

3.2.8.2 Customize Logo

The administrator can change the customized logo as following:

Company Logo is a png format file which contains company name. The recommendresolution is around 180*60Product Name Image File is a png format file which contains products name. Therecommend resolution is around 500*40Product Version Image File is a png format file which contains product ID or version.It will be following after Product Name Image. The recommended resolution isaround 150*20

3.2.8.3 Web Login Blocked IP

Here show the blocked IP for the web. When the system blocked an IP for attemptinglogin, you can unblock here. Click Web Service -> Web Login Blocked IP and thefollowing will appear:



Blocked IP The IP was blocked because of failed login

Login ID The ID was tried for the failed login

30


Login Time The time was blocked

Time to Unblock The time will be unblocked

3.2.9 Voice Logging

It is only available when voice logging license is activated and used to decide how tostore the voice logging file. Click System -> Voice Logging and the following willappear:



Recording File Format The recoding file format could be MP3 or WAV format.

MP3 Encoding MP3 encoding method, it could be CBR (Constant BitRate) or VBR (Variable Bit Rate) depending on thecompression ration and quality.

Voice Quality Selected voice quality when VBR is used. 0: best quality(220-260K), 9: best compress ration (45-85K)

Bit Rate The selected bit rate will be used when CBR is selected.

Recording FileKeeping Days

How long the converted recording file will be kept.

Joint Stereo Whether enable joint stereo or stereo mode for thecreated MP3 file.

iLBC Payload Type The payload type used for iLBC in order to decode it.

Using the System

31


iLBC Bit Rate The iLBC used bit rate (30 ms or 20 ms payload size)which will be used to decode

3.3 Monitor

The Monitor will display the current status (today) of SIP call traffic including traffic,quality, quantity, ASR, call status etc. Click different date from the calendar, theadministrator can see the specified date's call status. The calendar is look asfollows:

Select the specified date to see the summary of call traffic, status and quality. Thefollowing is the each summary's description.

32

3.3.1 Call Traffic

The Call Traffic shows the call attempt for each hours as follows:

3.3.2 Call Quality Split

The Call Quality Split show the call quality split for the specified date under thedefault jitter buffer simulation. You can select different simulation strategy from thetop right selection and see different result based on different assumption. Normally, ifyou have many dissatisfied or nearly all dissatisfied, it indicates your network or callquality had problems.

Using the System

33

3.3.3 Call Quality Trend

The Call Traffic Trend shows the average MOS value for the specified date underthe default jitter buffer simulation. You can select different simulation strategy fromthe top right selection and see different result based on different assumption.

34

3.3.4 Call Status Split

The Call Status Split show the call connected status distribution per hour. You canverify the reject reason is normal or not.

3.3.5 Average Success Rate

The Average Success Rate shows the connecting rate per hour. It is useful toidentify whether your VOIP network had problem or not by simply check the averageconnecting rate.

Using the System

35

3.3.6 Average Call Duration

The Average Call Duration shows the average talk time per hour. It can also be areference indicator to be verified for your voip network. It is less than your usualduration, it could be considered the service had problem.

36

3.3.7 Call Distribution

The Call Distribution shows that IPV4 and IPV6 call distribution. The administratorwill able to know the IPV6 migration status.

Using the System

37

3.3.8 Codec Split

The Codec Split show the distribution of each audio codec. The current recognizecode is G.711 Alaw, G.711 Mulaw, G.729A, GSM, iLBC.

38

3.3.9 Problem Calls

The Problem Call show the potential calls which might has problems. The type ofproblem calls include One Way Voice calls, screened calls, low voice quality calls,potential calling attack calls and long pdd calls count for administrator.

Using the System

39

3.3.10 Post Dial Delay

The Post Dial Delay shows each hours' average post dial delay time. The longertime of PDD indicate the longer wait after user dialed the number.

Using the System

41

3.3.11 Video Codec Split

The Video Codec Split show the distribution of each video codec.

3.4 Calls

The Calls will display each call's detail including sending/receiving packages, callquality, delay, jitter and some other detail to help system administrator to identify itsvoip network issues. Click Calls and the following screen will be displayed.

42

Select the date on the calendar to check each day's calls. And the following quickpaths will make the life easy to see the problems calls.

Click Call to see whole calls for the selected day.

Click Low Voice Quality Calls to see those calls' MOS value is lower than LowVoice Quality MOS Threshold in SYSTEM -> Quality Thresholds.

Click Potential Calling Attack Calls to see whether had any potential sip calls orabnormal calls.

Click One-way Voice Calls to check whether had one way voice calls or not.

Click Screened Calls to check whether had any calls attempts to the screened callsor not.

Click Long PDD Calls to check those calls had post dial delay longer than PDDthreshold.

Click Blocked UA Calls to check those calls are coming from blocked user agentswhich normally is coming from SIP attack hacker.

Click Loop Calls to find out those call which might have potential looped call. Thesearch is based on the Universal call ID and a programmable loop threshold. It willfind one UCID which have more than defined threshold counts.

Using the System

43

3.4.1 Search Criteria

The Calls can be searched based on the following search conditions.

After input the search condition, the following result will be display:



Time Search Time could be the following format:yyyy/mm/dd yyyy/mm/dd hh:mm:sshh:mm:ss

Duration Call duration

Telephone Number The telephone number to be filtered.

IPV4 Address The IP v4 address used to be filtered

IPV6 Address The IP v6 address used to be filtered

Fixed 50 ms MOS The MOS value based on the fixed 50 ms jitter buffersimulation

44


Fixed 200 ms MOS The MOS value based on the fixed 200 ms jitter buffersimulation

Adaptive 500 msMOS

The MOS value based on the adaptive 500 ms jitter buffersimulation

Average Jitter The Average Jitter used to be filtered

Max Jitter The Max Jitter used to be filtered

Burst Loss Rate (50ms)

The burst loss rate based on fixed 50 ms jitter buffersimulation

Burst Loss Rate (200ms)

The burst loss rate based on fixed 200 ms jitter buffersimulation

Burst Loss Rate(Adaptive)

The burst loss rate based on adaptive 500 ms jitter buffersimulation

Packet Loss Rate The RTP packet loss rate (loss packets v.s. receivedpackets)

Response Code SIP last received response code for the call

One-Way Voice Call Filtered those One-way voice calls

Payload Audio Codec to be filtered

SIP Call ID SIP Caller ID to be searched

PDD Time Post Dial Delay time

SIP Proxy IPV4 SIP proxy IPV4 address


Connect Time The connect time of search called

Save Time The save time of calls

Capture Interface The capture interface

See BYE Whether the call was connected and seen BYE or not

SIP Call ID SIP Call ID

Universal Call ID Universal Call ID for tracking purpose

Using the System

45

3.4.2 Calls Detail

The Calls will list the filtered detail record as follows:

Indicate this call is a Video call

Click to download the captured packets.

Click to download universal UID call which might combine more calls into 1 filefor tracking purpose.

Click to play the converted voice.

Click to play the voice logging file.

Click to see the detail call information and voice quality as follows:

Each column of Calls Detail are described as follows.


Call Time Calling Time

46


Caller Calling Party Number

Called Called Party Number

Caller/Called Payload Caller/Called used audio codec

Caller IPV4/IPV6 IPV4 and IPV6 address of caller party

Called IPV4/IPV6 IPV4 and IPV6 address of called party

Caller/Called Device SIP User Agent

Response Code/String Final SIP response code and string

SIP Call ID SIP call ID

Universal Call ID Universal Call ID for tracking

Post Dial Delay Post Dialed Delay Time

SIP Proxy IPV4 SIP Proxy IP V4 Address

SIP Proxy IPV6 SIP Proxy IPV6 Address

Save Time The time to save this call

Connect Time The time to connect this call

Ring Time The time to ring this call

Disconnect Time The time to disconnect this call

Capture Interface The interface to capture this call

Save Capture File Whether save the captured data or not

See BYE Whether the call was connected and seen BYE or not

Fax Detected Whether this call start the T.38 fax or not?

Using the System

47


Video Detected Whether this call start the video call or not?

Video Payload If this call is a video call, this is the video payload theyare negotiated.

TCP Protocol Whether this call is using SIP TCP or not?

RTP Tangled Whether RTP is tangled or not?

Record File Status The converting state of voice logging (only availablewhen voice logging license is granted)

Convert Time The voice logging file converting time (only availablewhen voice logging license is granted)

The caller and called party predicated MOS value based on different jitter buffersimulation.

The caller and called party packet loss rate based on different jitter buffer simulation.The most important loss is Burst Loss Rate which might make the voice quality badcompared to general loss rate.

48

The caller and called party packet delay distribution. You can see how many packetswere delayed based on different delay time.

The caller and called packet loss rate.

This is the packet lost count. More packets lost at the same time will affect voice

Using the System

49

quality more.

3.5 Register

The system can monitor SIP Register message (additional license is required) andsee whether had potential register timeout, register failure or attack etc. Click onRegister the following navigating tree will appear:

Select the date on the calendar to check each day's register. And the followingquick paths will make the life easy to see the problems register or current register

50

status.

Click Register Status to show the current user register status.

Click Register Expires to show those had register timeout's message.

Click Potential Register Attack to show those potential attack of register.

3.5.1 Search Criteria

The system provides search to find out which register you would like to investigatemore. Click Search button and the following screen will appear:

The detail of each search parameter is described as below:


Register Time Search register time could be the following format:yyyy/mm/dd yyyy/mm/dd hh:mm:sshh:mm:ss

Save Time The save time of register

User The register user

Register Status The Register Status could be the following:Registered: The user was registeredRefresh Save: The user was registered and saved forverify based on saving time.Unregistered: The user was not registered or unregisteredTimeout: The register message was not send in time torefresh the register

IPV4 Address The IP v4 address to be matched

Using the System

51


IPV6 Address The IP v6 address to be matched

Response Code SIP last received response code for the call

SIP Call ID SIP Caller ID to be searched



Capture Interface The capture interface

Contact URL The register contact URL

3.5.2 Register Detail

The register message list are describe as below:

Click to download the captured packets.The last or current status was show as

.

Click to see the detail call information of register as follows:

Each column of Register Detail are described as follows.

52


Register Time The time to register

User SIP register user

Register Status The final register status of the capture

Caller IPV4/IPV6 IPV4 and IPV6 address of CPE

SIP Proxy IPV4/V6 IP address of SIP proxy

Device SIP User Agent

Contact URL SIP Contact URL

Response Code Register Response Code (0 mean no reply)

Responding String Register Responding String

SIP Call ID Call ID of this call

Expires The negotiated register expires time

Save Time The time to save this call

Capture Interface The interface to capture this call

3.6 Alert

The system alert can be programmed to fire alerting message to administratorthrough email or syslog for such as low voice quality, potential SIP attack, low ASRetc.

3.6.1 System Alert

The system can be set to automatically send the system alert to administratorthrough syslog or email. It is only available when the system running stand-alone.Click SYSTEM -> System Alert to view and change the settings.

Using the System

53


ParameterName

Description

System AlertThreshold

The filter level to send the alert out. The default is level of"Notice".

Alert to Syslog Whether to send the system alert to syslogd server or not.

Syslog Receiver IP The syslogd server to receive the system alert.

Alert to Email Whether to send the system alert to the listed email accountor not.

SMTP Server The SMTP server for sending the system alert mail notice.

Email From The email sending account (FROM)

Email To List The email addresses to receive the system alert email.

Email Subject The email subject for the system alert notice. The variable "$HOSTIP$", Host IP address, could be used in the subjectto make the subject easy to be read (e.g. System AlertNotice from $HOSTIP$).

Email User ID The email sending account ID

Email UserPassword

The email sending account password

54

3.6.2 Alert Criteria

Alert Criteria is used to set the condition of alerting. If the condition was matched,an alert will be send to administrator through syslog and email depending on thesettings of system alert. The system will check the system status in 5 minutesinterval. Click ALERTS -> Alert Criteria and the following screen will display.

Click New or Modify the customized criteria.

3.6.2.1 Low ASR Alert

Low average success rate (connected rate) sensor can be used to detect if youroutgoing call trunks had problems or not. Select the alert type to Low ASR and thefollowing screen will display.

Each column was described as following:


Criteria ID The identification of this criteria ID

Using the System

55


Alert Type The sensor is used to detect the alert

Checking Interval How long ago call information will be verified in 5minutes interval. If you set to 15 minutes, each time (5minutes interval) the system will check 15 minutesago till now's call information to detect the alert.

Criteria Content Send alert if average success rate is lower than thepredefined ratioand total call attempt is greater than the predefinedcount.

The fired alert will be as follows:

Low ASR Alert: ASR is lower than 10%(0%) of total calls (18) during the period from2011-06-13 14:00:00 to 2011-06-13 14:10:00

3.6.2.2 Low Voice Quality

Low voice quality sensor can be used to detect if your network had low voice qualitycalls or not. Select the alert type to Low Voice Quality and the following screen willdisplay.





56



Criteria Content Send alert if low voice quality Satisfied calls basedon Fixed 50 ms Jitter simulation exceed 2% of totalconnected calls and total connected calls are greaterthan 20


Low Voice Quality Alert: 0.2% calls (3) of total calls (1000) have low voice quality(MOS <3.0) based on 200 ms jitter simulation during the period from 2011-05-0117:25:00 to 2011-06-01 17:30:00

3.6.2.3 High Abnormal Calls

High Abnormal Calls can be used to detect if your SIP calls had a specified problemsuch as too many temperate unavailable (480) or forbidden (403) etc. Select thealert type to High Abnormal Calls and the following screen will display.






Using the System

57


Criteria Content Send alert if calls terminated by reason 404 exceed10% of total calls and total calls are greater than 10


High Abnormal Calls Alert: 55% calls (10) of total calls (18) belongs to abnormalcalls during the period from 2011-06-13 14:40:00 to 2011-06-13 14:45:00

3.6.2.4 Low Call Duration

Low call duration sensor can be used to detect if your network had short call durationor not. Select the alert type to Low Call Duration and the following screen will display.






Criteria Content Send alert if calls duration is lower than 60 secondsand total connected calls are greater than 5


Low Call Duration Alert: Call Duration is less than 60 seconds (total connected calls:5) during the period from 2011-06-13 14:00:00 to 2011-06-13 14:10:00

58

3.6.2.5 High Delay Packets

High Delay Packets sensor can be used to detect whether you voip calls hadnetwork delay or not. Select the alert type to High Delay Packets and the followingscreen will display.






Criteria Content Send alert if calls package delay count of 200 ms (orhigher) is greater than 10 packets and totalconnected calls are greater than 10 calls


High Delay Packets Alert: There are 12 calls have more than 10 packets of 200msdelay during the period from 2011-05-01 17:25:00 to 2011-06-01 17:30:00

3.6.2.6 Screened Calls

Screened Calls sensor can be used to detect if your users is calling those screenednumber e.g. international calls or not. Select the alert type to Screened Calls and thefollowing screen will display.

Using the System

59






Criteria Content Send alert if calls to the screened destination isgreater than 5 calls.


Screened Calls Alert: There are 7 calls are trying the blocked destination during theperiod from 2011-06-15 15:40:00 to 2011-06-15 15:45:00

3.6.2.7 High Packet Loss Rate

High Packet Loss Rate sensor can be used to detect whether you voip calls hadnetwork jitter or loss or not. Select the alert type to High Packets Loss Rate and thefollowing screen will display.


60





Criteria Content Send alert if calls which Burst Loss Rate is greaterthan 1.0% based on Fixed 200 ms Jitter simulationexceed 10 calls


High Packet Loss Rate Alert: There are 2 calls has high loss rate (0.001000) Jitter(50) during the period from 2011-06-13 21:50:00 to 2011-06-13 21:55:00

3.6.2.8 Potential Calling Attack

Potential Calling Attack sensor can be used to detect if there is a SIP attack istrying to access or try your system or not. Select the alert type to Potential CallingAttack and the following screen will display.





Checking Interval How long ago call information will be verified in 5minutes interval. If you set to 15 minutes, each time (5minutes interval) the system will check 15 minutes

Using the System

61


ago till now's call information to detect the alert.

Criteria Content Send alert if potential SIP attack call attempt isgreater than 5 calls.


Potential Calling Attack Alert: The IP address 2001:c08:3700:ffff::f:ae55 has toomany failure calls (6) during the period from 2011-06-15 15:50:00 to 2011-06-1516:20:00. It might be come from a SIP attack.

3.6.2.9 Potential Register Attack

Potential Register Attack sensor can be used to detect if there is a SIP registerattack is trying to access or try your system or not. Select the alert type to PotentialRegister Attack and the following screen will display.






Criteria Content Send alert if abnormal register attempt is greater than5 (for example)

62


Potential Register Attack Alert: The IP address 112.104.95.153 has too many failureregister (1) during the period from 2012-05-16 17:00:00 to 2012-05-16 17:05:00. Itmight be come from a SIP attack.

3.6.2.10 One-Way Voice Calls

One-Way Voice Calls sensor can be used to detect if your voip calls had one wayvoice happened or not. Select the alert type to One-Way Voice Calls and thefollowing screen will display.






Criteria Content Send alert if one-way voice calls is greater than 10(filter those no RTP calls: Enable). Normally, thosecall which RTP is not come through should be filtered.


One-Way Voice Calls Alert: There are 15 one-way voice calls of total connectedcalls (30) during the period from 2011-05-01 17:25:00 to 2011-06-01 17:30:00

3.6.2.11 Long Post Dial Delay

Long Post Dial Delay sensor can be used to detect if there is a SIP calls which hadlong post dial delay (over the setting of PDD threshold in Quality Thresholds). Selectthe alert type to Long Post Dial Delay and the following screen will display.

Using the System

63






Criteria Content Send alert if long post dial delay calls is greater than5.


Long Post Dial Delay Alert: There are 6 calls which Post Dial Delay is longer than 0seconds during the period from 2012-05-16 16:15:00 to 2012-05-16 16:20:00

3.6.2.12 Register Expired

Register Expired sensor can be used to detect if there is a SIP end point lose itsregister refresh or not. Select the alert type to Register Expired Alert and thefollowing screen will display.

64






Criteria Content Send alert if register expires found.


Register Expired Alert: The Extension <20021> is not registered anymore. RegisterTime: 2012-04-10 18:04:51, Expires: 60, Device: eyeBeam release 1102q stamp51814

3.6.2.13 Blocked UA Calls

Blocked User Agent Calls sensor can be used to detect if there is any potential SIPhacker is attacking your SIP server or not. Select the alert type to Screened Callsand the following screen will display.





Using the System

65



Criteria Content Send alert if calls to the screened User Agent isgreater than 5 calls.


Blocked UA Alert: There are 1 potential SIP Attacks (0) which are coming fromblocked UA (sipcli) during the period from 2016-05-20 10:30:00 to 2016-05-2010:31:00

3.6.2.14 Loop Call Alert

Loop Call sensor can be used to detect if there is any potential loop call or not. Select the alert type to Loop Calls Alert and the following screen will display.






Criteria Content Send alert if calls attempt per UCID is greater thanthreshold xxx.

66

The fired alert will be as follows:Loop Call Detected. Universal Call ID (xxxxxxxxx) has xxx calls which is over loopthreshold 70 during the period from 2020-07-03 15:51:00 to 2020-07-03 15:54:00

3.6.3 Screened Number

Screened Number is used to be display or detect those call are not allow to becalled such as international calls. Click ALERTS -> Screened Number and thefollowing screen will display.

Click New or Modify to have the following screen.

Each column was described as follows:


Screened Prefix The called prefix should be screened

Screened PrefixName

The name of the screened prefix

Using the System

67

3.6.4 Screened User Agent

Screened User Agent is used to be detect those calls which might be coming froma SIP hacker or attacking tool. The system will create some screened user agent forthose well know SIP attacker. Click ALERTS -> Screened User Agent and thefollowing screen will display.

Click New or Modify to have the following screen.

Each column was described as follows:


User Agent The SIP User Agent prefix should be screened

Description The description of the screened User Agent

3.6.5 Report

The fired quality alert can be queried here. Click ALERTS -> Report to have thefollowing screen.

68

The search condition is described as follows.


Time The alerting time

Type The type of quality alerting

Level The alerting level

Content The keyword to be searched in the content

3.7 Report

The system provides the quality statistic report for voice quality summary report,quality performance report and quality call split report based on system wide,customer based or provider based. Click REPORT and the following reporting treewill appear:

Click the report you would like to show and provide the search condition to generate

Using the System

69

the report you need.

3.7.1 Quality Summary Report

The quality summary report show the voice quality summary. Click REPORT ->System, Customer or Provider Quality Summary Report and the following willappear:

Input the search condition and click Apply. The following will appear:

For System Quality Summary Report:

For Customer Quality Summary Report:

For Provider Summary Quality Report

70

The detail of each field as described as follows:

Field Name Description

Customer ID The ID of the customer

Provider ID The ID of the provider

Time Time of the reporting period

Calls Number of calls within this period

ASR Average success rate within this period

Duration Total call duration of this period

One-Way Voice Number of one way voice calls

Screened Calls Number of screened calls

Low Quality Calls Number of low quality voice calls

Attack Calls Number of SIP attacking calls

Long PDD Calls Number of long PDD calls

Using the System

71

3.7.2 Quality Performance Report

The quality performance report show the voice quality performance for each hour ordaily basis. Click REPORT -> System, Customer or Provider Quality performanceReport and the following will appear:


For Customer Quality Performance Report:

For Provider Quality Performance Report

72








Calculated Calculated calls for voice quality

Caller MOS Average MOS value of caller

Called MOS Average MOS value of called

Very Satisfied Number of very satisfied voice quality calls

Satisfied Number of satisfied voice quality calls

Some Dissatisfied Number of some dissatisfied voice quality calls

Many Dissatisfied Number of many dissatisfied voice quality calls

Nearly All Dissatisfied Number of nearly all dissatisfied voice quality calls

Using the System

73


Terrible Quality Number of terrible voice quality calls

3.7.3 Quality Call Split Report

The quality call report show the split of call distribution. Click REPORT -> System,Customer or Provider Quality performance Report and the following will appear:


For Customer Quality Call Split Report:

For Provider Quality Call Split Report

74






Calls Number of calls within this period



IPV4 Number of IPV4 calls

IPV6 Number of IPV6 Calls

G.711A Number of calls are using G.711A codec

G.711U Number of calls are using G.711U codec

G.729A Number of calls are using G.729A codec

Using the System

75


GSM Number of calls are using GSM codec

iLBC Number of calls are using iLBC

404 Number of calls had 404 final response





3.8 Administration

The Administration setting includes the user account management, clear historicaldata, start or stop the service and web access log as follows:

3.8.1 Service Control

Click ADMINISTRATION -> Service Control to start the voice quality monitor ornot.

76

Click Restart All Service button to restart all sniffer interfaces.Click Reboot System button to restart the server.

3.8.2 Clean-Up

Click ADMINISTRATION -> Clean-Up to clear unnecessary DB records and files.

Select the periods of data to be cleared and click Clean button to clean thoseselected data.

3.8.3 Account

Click ADMINISTRATION -> Account to view the current settings of Administratoraccount. The following screen will appear.

Click New to add a new user and the following screen will appear.

Using the System

77



User Mode Activate or de-activate the user

User ID The user ID to login

Password The user password

Authorization The access right will be giving to this user. Administrator canhave full access right while supervisor can be limited basedon grated access control rights.

Language The web GUI language when the user login.

For supervisor, Administrator can assign the following rights by click AccessControl button as follows:


System ParameterModification

Whether this user can modify system related parameters ornot?

Interface Control Whether this user can control interface for start, stop orrestart?

78


Capture FileDownload

Whether this user can download the captured files or not?

Voice FileDownload

Whether this user can download/play voice file or not?

3.8.4 Web Provisioning

The system will record down all the access to the system from web. Theadministrator can use it to audit the system and tracking the changes. Click REPORT -> Web Provisioning to view the report as follows:

The detail of each report field is described as follows:


Time The time to access web

Target The web target to be accessed

Operation The operation madden by user

Modifier The user who made the change

Using the System

79


Login IP IP Address to Access the Server

Update Value The value changed detail

Select one of record and click the Detail button. The following detail for such recordwill appear.

3.8.5 Upgrade System

Use Upgrade System to do the application patch by clicking ADMINISTRATION -> Upgrade System. Please only use the certificated patch file to do the upgrade.Otherwise, it will had problems.

After upgrade, reboot the machine to take effective.

3.8.6 Downgrade System

Use Downgrade System to restore the application to previous patch by clickingADMINISTRATION -> Downgrade System.

Select the downgrade file name which is backup by the upgrade automatically. ClickApply to downgrade. After downgrade, restart the service to take effective.

80

3.9 Commit

After you change the system settings, you need to apply it by clicking the COMMITand the following popup screen will appear:

Select OK to commit the changes.

Appendix

81

4 Appendix

Enter topic text here.

4.1 List of Used Network Ports

The following is the list of used TCP/IP ports. The network administrator can use it toset the firewall when necessary.

DefaultPorts

Protocol

Description Configuration Path

9320 TCP HTTP port for administrator

SYSTEM -> WEB Service -> HTTP ServicePort -> AdministratorOnly be opened in firewall when necessary.

9321 TCP HTTPS port foradministrator

SYSTEM -> WEB Service -> HTTPSService Port -> AdministratorOnly be opened in firewall when necessary.

EZQoS-3200 VOIP Monitor Administrative Guide Administrative...Getting Start 7 2 Getting Start After...

Documents

Transcript of EZQoS-3200 VOIP Monitor Administrative Guide Administrative...Getting Start 7 2 Getting Start After...