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FAQs covering :-

  • VoIP Monitoring and Troubleshooting

 

FAQ Index 

Click to view FAQs for listed subject

 

Click to view FAQs for listed tools

 

Toolbox and access information

 

 

 

Part of

Facilities accessible via autoVoIP tree branch and autoVoIP Call Logging tree branch.

 
Video Tutorial information

These Tutorials can be used to answer questions:-

  • How to Monitor and Troubleshoot VoIP Networks - Part 1 and 2

  • How to install, configure and use SQL call Logging tool

  • How to use Frame Flow Analyzer

  • How to deploy Codima Toolbox

  • Reports Manager - Overview and Applications

  • How to use Reports Manager

  • How to use Codima Toolbox Alarm System

Click here to access Tutorial Sign up access point

 
  Network and Deployment Information

Remote Manager - Probes

For use on the following VoIP Networks:-
  • SIP
  • Cisco Skinny (SCCP)

For deployment information, see the :-

  • VoIP Monitoring and VoIP Troubleshooting tools section.
  • Passive Analysis and Active Operations section.

of the Codima Toolbox Deployment Guide

 

The Remote Manager tool is included in all the Codima Toolboxes, it is used to view and in some cases control Remote systems. These remote systems can be:-

  • Probes - Which enable you to use the functions accessible via the remote systems autoAnalyzer and autoVoIP tree branch. Extending  the domain being monitored by the Passive Analysis tool.

For more information, click on options below:-

 

FAQs covering Benefits

 

 

 

FAQs covering Installation, Licensing and Deployment

Installation :-

How much memory do I need to install?

How much disk space do I need?

Licensing :-

For more information on the license process see :-

Deployment :-

For other installation issue - see FAQs for autoAnalyzer tree branch

 

 

FAQs covering Scalability
 

 

 

FAQs covering Software Delivery, Software Updates and Software Release Contents

Software Delivery and Software updates:-

Software Release content and schedule :-

  • How do I find out what is included in the latest software releases?
  • How do I find out what is included in older releases?
  • How do I get the latest software version?
  • How do I get a copy of the Codima Toolbox software?
  • How do I check the version number of the software I have installed?

See  FAQs covering Software Releases

 

 

FAQs covering Software Delivery, Software Updates and Software Release Contents

Software Delivery and Software updates:-

Software Release content and schedule :-

  • How do I find out what is included in the latest software releases?
  • How do I find out what is included in older releases?
  • How do I get the latest software version?
  • How do I get a copy of the Codima Toolbox software?
  • How do I check the version number of the software I have installed?

See  FAQs covering Software Releases

 

 

FAQs covering Troubleshooting and Check lists
Troubleshooting :-

- Frame capture (to supply evidence) :-

Check lists:-

 

For more on Check lists/Reference Material and Evidence requirements - see FAQ - Troubleshooting Codima Toolbox

 

 

FAQs covering Tools

 

autoVoIP tree branch provides access to functionsQuesti
Area / Question
Tutorials :-
 

 
Getting Started :-
 

 
Applications/Scope/Operations :-
Applications :-

Scope/Operations :-

- Automatic restart :-

- Performance/Network type :-

- Protocols

- Phone/Codec support :-

- Call Analysis/VoIP Call Group Analysis : Statistics gathered by Group (CAG)

- Tracking RTP (Media)

- Tracking Phones and Registrations :-

- VoIP Troubleshooting :-

Uses - Troubleshooting Report

- Reports:-

- Remote operation :-

- SLAs/QoS :-

- Miscellaneous :-

 

 

 autoAnalyzer/Review Frames tree branch provides access to frame file library - Open Frame file and Right Click to access.

Applications :-

Scope/Operations :-

Flow Analyzer/RTP Analyzer/Call Playback :-

         The Flow Analyzer is also the access point for the following facilities

Can I use Wire Shark™ (formally known as Ethereal™) trace files?

Uses - Flow Analyzer and Call Playback facility

Uses - Call Playback facility

  • Can I analysis the RTP Payload? - e.g.,
    • Identify Equipment queues, by showing you where the frames are bunched together.
    • See the frame drop pattern
    • See if multiple paths are present by looking for out of sequence frames. 

    Uses - RTP Analyzer

 

 

 

Applications :-

Scope/Operations :- Alarm facilities

 

 

 
Applications :-

Scope - Statistics Reports :-

 

autoVoIP Call Logging tree branch provides access to functions

Area / Question
Applications :-

Scope/Operations :-

SQL Call Logging tool

 

 

 FAQs on miscellaneous/linked subjects
Area / Questions
VoIP Technology :-

 
Microsoft® Patch level :-

 

 

 

 FAQs and answers

Tutorials

Are there any tutorials to help me get started?

Yes - there are tutorials to assist you in learning how to use the facilities available via the autoVoIP tree branch. They are accessible from the Help facility and from the Codima Website.

Click here to access the sign in page for the Tutorials:-

The tutorials are titled :-

  • How to Monitor and Troubleshoot VoIP Networks - Part 1
  • How to Monitor and Troubleshoot VoIP Networks - Part 2

There are also tutorials covering the Frame Flow Analyzer and the SQL Call Logging Add on

  • How to use Frame Flow Analyzer
  • How to install, configure and use the SQL Call Logging tool

 

 

Installation

 

Can I install Toolbox on a Virtual Machine?

 

Yes, a supported operating system would need to be run inside the Virtual Machine. You also need to ensure that you have allocated sufficient disk space and memory to run the toolbox application. Failure to do this can cause operational problems.

 

Click here for details covering supported operating system, disk space and memory requirements.

 

 

Can the Toolbox operate using a Wireless card for packet capture?

 

The NICs need to go into promiscuous mode to capture frames and most wireless NIC do not support this.

 

 

Can I run the Toolbox as a Windows Service?

 

No, the Codima Toolbox won't run as service. If you do this you can't see the interface to interact with so you have to be logged in. Users can make the Toolbox start at windows startup by putting the shortcut to it in the startup programs folder.

If users are running Codima tools that need 24/7 operation, then you must run the application continuously and not log out.

 

Can I install the Codima Toolbox on the same host platform as the Codima Spider?

You can either install a Codima Spider Server or a Codima Toolbox. However you can have a Codima Toolbox with Web Access, that can act as a Codima Spider Server but still allow access on the Host Platform to the tools in the Codima Toolbox.

 

 

Can I use a Host Platform with a 64bit Operating System?

What Operating Systems can I use on the Host Platform?

 

The Codima Toolbox will operate on platforms running the following Operating Systems

  • Microsoft® Windows Vista, XP, Server 2003, Server 2008, Server 2008 R2 or Windows 7 with 2GHz CPU (minimum)

 

Toolbox includes WinPCap drivers (used for frame capture) that support 64bit Operating systems.

 

 

What are the Host Platform requirements for the Toolbox?

  • How much memory do I need to install?
  • How much disk space do I need?

Click here to access information on the Host Platform requirements for the Codima Toolboxes

 

See also FAQ

 

 

 

What configuration is required to monitor VoIP traffic?

Configuration - Network Adapters (System Module)

Ensuring that the Toolbox is correctly deployed to enable it to passively monitor the VoIP Media, Signaling and Reporting frames is important and may require you to configure the network adapters assigned to the Toolbox platform.

A standard Toolbox can operate with three Network Adapters simultaneous, each one being configured for connection to a specific port on the Host Platform. For deployment guidance, see titled:-

The Host Platforms Network Adapter(s) must be correctly configured and connected to the network to monitor traffic.

Multiple port configuration

The following help entry provides configuration instructions when you need to change the default settings. The default setting is to configure the first suitable network adapter it finds to Port 1 (analysis) and the next one to Port 3 (comms) and if it just finds one suitable network adapter it allocates it to both Port 1 and the Comms Port.

Configuration - functions accessible via autoVoIP tree branch

autoVoIP/Configuration/General tree branch

The Toolbox has the following configuration capabilities,  they fall into two categories :-

  • User - Used to configure "user requirements", i.e., issue that are specific to the network being monitored.
  • System - Used to configure the Toolbox itself.

These configuration features are for use by expert users who have detailed knowledge of the network being monitored. Detailed information is provided in the Toolbox help file.

Example showing interface used to configure VoIP SLA.

 

How do I configure the Host Platform to connect to multiple ports using multiple Network Adapters?

The following display shows the actions need to configure Ports 2 and 3.

 

 

Licensing

How is Toolbox licensed?

A standard Toolbox license will control the number of Registered phones that the Toolbox is allowed to monitor.

A PAYG license will not set limits, but user will be charged for the number of Registered phones the Codima Discovery Engine discovers during each month.

Other license issues

All Tools in the Codima Toolbox are required to be licensed, the license files cover the following :-

  1. Toolbox License (File name = LICENSE.TXT)

The Toolbox License controls the following:- 

  • Toolbox structure, e.g., access to autoVoIP tree branch

  • Discovery Device limits, e.g., 50 Managed Devices

  • VoIP phone limits, e.g., 100 SIP Phones

  • Expiry date

  • MAC Address used to machine lock the software

  • Discovery Engine Version, e.g., 3.0-CDE

  • Software Release Version, e.g., 4.00 0001

  • The Demonstration/Evaluation status

  • The PAYG status, i.e., user has a standard license or a PAYG license.

  1. SNMP License (File name = LICENSE.KEY) 
This is the license file for the SNMP module, all Codima Toolboxes are required to have this license.  
 
Error Message 
If you do not install the License.Key file, you will get a regular warning message reporting that you are using an evaluation version of the WinSNMP module 

 

How do I upgrade my Phone Count License?

Contact your Codima Toolbox supplier to obtain a License that allows for more phones. There will be a cost associated with this upgrade as the standard Toolbox licenses for the functions accessible via the autoVoIP tree branch are based on the phone count. You can also consider using a PAYG license.

 

The Troubleshooting dashboard will provide phone counts and show limits:-

 

 

 

How do I find the MAC Address of the Platform I wish to install on? - required to machine lock license

 

The license files used by the tools in the Codima Toolbox are linked to the MAC Address of the Platform you install the software on. To obtain the MAC Address of a Platform you should type  ipconfig/all from a DOS window.

 

The MAC Address is a 12 digit hex number, e.g., 00-07-E9-5A-77-DB

 

To get to a DOS window, click Start, then Run, then type cmd in the text box.

If Platform has multiple MAC Addresses, you need only to supply one for the license link.

 

DOS Window example:-

 

Alternatively if you are using a demonstration system, you can make use of the automated facility to request a license upgrade, this facility automatically finds your MAC address.

 

Display showing interface used to request license upgrades:-

  

 

 

 

What is a PAYG License?

Toolbox PAYG (Pay As You Go) licenses are provided to users who are being charged by usage. Usage applies for example to the number of managed devices counted during the operation of the Codima Discovery Engine, the number of Registered Phones, or the loading level used to stress test the network.

When a user has a PAYG license, it is important that they are connected to the Internet, to ensure that the Registered Phone and the managed device counts can be uploaded to the Codima PAYG servers and monthly usage reports created.  These reports are then used to bill clients.

 

 

Scalability

Are there any limits to the number of phones I can monitor with Toolbox? -

(VoIP Monitoring & Troubleshooting Toolbox and Codima All in One Toolbox)

 

There are two aspect that can set limits:-

  1. The License - When you have a standard Toolbox license (as opposed to a PAYG license), see FAQs:-

     

  2.  The Host Platform - see FAQ :-

However both these limitations can be addressed to make the Toolbox scalable for use on large networks. You can upgrade your license to support more phones and you can add additional Probes to extend the domain monitored. See FAQ :-

 

 

Can I have some guidance on the number of Toolboxes needed to monitor my network?

- scaling the deployment to handle monitoring larger networks

 

The solution required to monitor larger networks would need to include at least one Toolbox and a number of Probes

 

Diagrams showing possible deployment positions:-

 

 

The Toolbox deployment requirements will be subject to the available monitoring points on the network, for example you will need to establish:-

  • How you are going to connect each Toolbox/Probe into the network, e.g., using Switches (with Span Ports),  Hubs and/or Intelligent Taps - this will be based on where on the network it is possible to tap into the relevant traffic flow (i.e., Skinny and RTP/RTCP or SIP, SDP, RTP/RTCP frames)

Connection using Intelligent Taps

• Where the Intelligent Taps are physically located.

• What the capabilities are of the Intelligent Taps and how the Taps are to be configured to enable the Toolbox to monitor the Signalling and RTP/RTCP traffic flows.

Connection using mirror/span ports

• Where the mirror/span ports are physically located.

• What the capability of mirror/span ports is, for example does it support RSPAN or VSPAN

• How the span ports are to be configured to enable the Toolbox to monitor the Signaling and RTP/RTCP traffic flows.
 

When deciding on the monitoring points/span port configuration you need to take into account the following:-

  1. The number of Phones/Call a single Toolbox/Probe can process, see FAQ:-

  1. The capabilities of the Intelligent Taps

  2. Mirror/Span ports limits on the volume of traffic they can mirror, so that limit combined with the phone/call limits for a single probe will enable you to judge where you can locate probes.

  3. The type of Mirror/Span port, for example if the switch supports VSPAN you can potentially have a single probe monitoring traffic from multiple VLANs

  1. The traffic flow on your network, see FAQ titled:-

 

 

How many Phones/Calls can a single Toolbox/Probe process?

Overview

The maximum concurrent calls that can be analyzed by the Toolbox/Probe depends on the Host Platforms specification and the protocol used to measure QoS, i.e., RTP or RTCP.

  • RTP - Each Voice call runs at about 50 RTP frames/second duplex - a typical Host Platform will handle around 1000 concurrent RTP calls.
  • RTCP - If we only monitor RTCP the maximum concurrent calls is very much higher at around 100,000. This is because there is low CPU processing overhead per frame and the packets are typically transmitted only every 5 seconds or longer.

The signaling protocol has minimum impact, processing the SIP packets for example although complex, is not a performance bottleneck as they are relatively infrequent, and Skinny signaling is a binary protocol so processing overheads are minimal.

Details

 

The Toolbox/Probe using a standard NIC and a suitable Host Platform will be able to process many thousands of Phones/Calls at Signaling Protocol Level. However if the Toolbox/Probe is also monitoring RTP then the count would be a lot less.

 

The following provides a rough guide:-

  • Monitoring Signaling only - 10,000 simultaneous calls

  • Signaling plus RTP - 1,000 simultaneous calls

 

Licenses are used to limit the number of phones logged on each system.


Calculation information:-

 

SIGNALING

 

Example - At SIP level, a call every 30 seconds with 8 frames per call =  packet rate for 10,000 phones is 10,000/30x8 = 2,666
frames per second.

 

A single Toolbox/Probe should be able to handle around 100,000 frames per second, so would not have any problems with this level of traffic.


MEDIA

 

Bottleneck can occurs when trying to process the MEDIA (RTP stream) at large concurrent call rates .

  • RTP - Each Voice call runs at about 50 frames/second

  • RTP/RTCP combined media stream - 50x2x10000 (see note) = 1,000,000 frames per second. - This would be too fast for a single Toolbox/Probe by a factor of 10, as a typical Toolbox/Probe platform will handle around 100,000 frames sec.

  • RTCP only - If we only monitor RTCP however it will be fine as the processing of RTCP frames is a low overhead per frame.

Note: Calculation is being applied to an example situation where there are 50 frames per second x 2 (bothways) x 10,000 phones.

In such cases, i.e., where call level is such that bottlenecks are likely to occur, you would use additional probes. Which makes the Toolbox system scalable for use on large networks, see FAQ

 

 

 

 

Is Toolbox scalable to handle a large number of phones on a distributed network?

 

Yes - For more information on scalability, see FAQs titled:-

 

Codima also provide a Technical Bulletin on this subject. Contact your supplier to obtain a copy, ask for:-.

  • Bulletin 1 - DISTRIBUTED ANALYSIS : VOIP MONITORING TOOL AND VOIP TROUBLESHOOTING TOOL

 

As well as the management system that allows you to view the analysis being undertaken by multiple Probes from a central point, you also have within a single system a number of simple methods to focus on specific phones or problems, there is no need to scroll through long lists of phones. For example each system has:-

 

A Troubleshooting Dashboard - this has two key areas to assist in fault isolation:-

  • The Search feature which can find Phones, by name, by ip address, by domain, or by phones descriptions, e.g., manufacturer, software level or by error type.

  • The Grid Filter Tabs which show only Active Phones or Phone reporting Errors or Phones with low QoS

Troubleshooting Dashboard Grid - Error Phones tab showing problem phone:-

 

A Phone Groups Manager Dashboard - the user can predefine different groups of phones based or IP address ranges or URL text strings. QoS, Call Counts and Codec Usage are tracked from the Groups, and the user can cycle through them in the Troubleshooting Dashboard display. This provides a method of focusing on a set range of phones.

 

 

A Phone Software Analysis Dashboard, which highlights errors and relates then to specific groups of phones, e.g. manufacturer/software level. 

Phone Software Analysis Dashboard example (SIP phones):-

 

 

How far back can I store phone history statistics for a large number of phones and how much disk space would I need ?

 

This is a complex question as there are many variables involved - some of which are network specific.

Key points

1. Each Toolbox* needs to allocates disk space to enable multiple History database files to store different sets of statistics - There are 30 database files in total - size range can be from 8MB to 1048MB for each file. So for maximum storage you can set for the database files used by the VoIP Monitoring tools is 1048Mb. The file will wrap when full, so you always have the last 1048Mb worth of statistics.

 

 

*When monitoring large networks, you may need multiple Toolboxes, see FAQ :-



2. Codima uses a proprietary high speed statistics storage system which allows it to collect bulk high resolution statistics e.g., minimum of 40 statistics for 500 phones in typically a few milliseconds on standard Host Platform.

3. For lower resolution statistics (phone statistics based on 15 minute intervals) - we can not give an exact figure, but typically a single Toolbox can undertake months of tracking for say 500 phones, before the file wraps.

4. For high resolution statistics (15 second intervals) - then the statistics associated with Phone Groups (rather than individual phones) could cover months if you are tracking say the statistics for 10 user defined groups, before the file wraps.

5. For providing call records, you also have the option of using the SQL Call Logging tool (Add on) - that contains detailed call information on a per call basis and maximum file size here would be terabytes.

 

For more on the SQL Call Logging tool, see FAQ :



6. You also have daily, weekly, monthly report facilities that will take information from history databases and create Word Reports. The report creation takes place at end of day, week, and/or month - allowing you to continually have trend report covering phone statistics.

It is not possible to provide a figure for the amount of space needed to store these word reports - that is subject to report size and report range activated. So for long term usage would recommend checking free disk space on a regular basis and removing older reports to make room for new ones. After running reports on the network for a couple of months, the user would get an idea of how much disk space a months worth of reports takes up.

 

For more information, see FAQ :-

 

 

 

 

 

 

Software Delivery and Software Updates

How do I get a copy of the Toolbox software need to operate the functions accessible via the autoVoIP tree branch?

Demonstration software

You can download trial Toolbox software from the Codima Web site download page - select the download for the VoIP Monitoring and Troubleshooting Toolbox.

Evaluation software or Purchased software

URLs to download software for the VoIP Monitoring and Troubleshooting Toolbox are provided in the installation instructions attached to the license delivery email.

 

How do I get the latest software version?

 

You should set up your Host Platform so that it is able to access to the Web. Then you can use the Automatic Installation Update facility. This will automatically tell you if there is an update to the system available. You will also need to be authorized to receive updates.

 

You can also use the Help Menu, where the option Check for Updates will open a Web page listing the available updates, which you can select to download.

 

For more information see Help entry titled:-

  • How to upgrade Software

 

 

 

Getting Started

Is there any guidance available to help me get started when using functions accessible via the autoVoIP tree branch?

Yes there is, use the following reference documents:-

  • Getting Started Guide - autoVoIP Tree branch -  URL to access this guide is provided when you download software and is also included in the license delivery emails when you evaluate or buy the VoIP Monitoring and VoIP Troubleshooting tools.

  • Operations Guide - autoVoIP Tree branch - Power Point Presentation, the URL to access this guide is provided directly to sales staff/resellers on request.

 

 

 

Applications

 

What are the Benefits of using the VoIP Monitoring and VoIP Troubleshooting tools?

 

There are a large number of benefits, here are some of them :-

  • Real time independent VoIP Server Analysis.

    • Can track Server Response times and identify if they are being effected by other network activities.

  • Real time independent Registration Analysis.

    • Registration Analysis by Server

      • Can provide security checks on which VoIP Servers are running on the network

      • Can show at a glance who is registered to which Servers

      • Can show failed Phone Registrations for the Servers.

      • Can find illegal phones attempting to access VoIP Servers

      • Can show when Phones have not received replies to Registration Requests – and subsequently do not get a dial tone

      • Can highlighting excessive Registrations

    • Registration Analysis by Phone

      • Can find misconfigured phones trying to register with the wrong Server IP Address.

      • Can find Register Timeout mismatches between the Server and the Phones

      • Can show Failed Registrations

        - where an error report has been monitored

        - where no response has been received from the SIP Server

      • Can show when phones are registering with more than one Server

      • Can show Phone Registrations Historically

  • Real time independent troubleshooting.

  • You can easily spot error patterns - Establishing:-

    • If only one phone is effected.

    • If multiple phones are effected .

    • What these phones have in common e.g., same manufacturer, same software level.

  • Can provide the information you need to use your resources efficiently - Identifying:-

    • If Error reports are intermittent or persistent.

    • If the problem is current - effecting the user now.

    • How serious the reports are by establishing when they occur.

  • Real time Service Assurance Monitoring for each Phone.

    • Automatically shows QoS by Phone, tracking jitter, lost frames, round trip delay, calculating R and MOS values.

  • Real time Service Assurance Monitoring at a Global level - Global QoS Analysis

  • Real time Service Assurance Monitoring for Phone Groups.

    • QoS Analysis for Phone Groups. Phone groups are user definable, so QoS analysis could be focused on a specific sector of the network, or a specific group of priority phone users.

  • Can easily track Call Loading and Codec usage Globally and by Phone Group.

  • You can do a very quick health check on the sector of the network being monitored.

    • You can easily see if you have patterns indicating that high priority large frames are clogging up the Switch/Router output Queues causing Jitter and Latency.

    • You can quickly see if the Voice VLAN has any non VoIP traffic on it.

  • You can be proactive :

    • QoS deterioration or increases in error reports for specific phones can be reported automatically via the Global Alarm System.

    • Increased Non VoIP traffic on Voice VLAN can be reported automatically via the Global Alarm System.

      • The Global Alarm System uses multiple alarm reporting methods (SNMP traps, emails, SMS text messages) and has an embedded email client that is independent of the Networks own email facilities.

  • Evidence at your finger tips.

    • Reports covering the history of phone group traffic patterns gathered using passive analysis can be automatically produced or produced on demand for selected time periods. They will enable you answer questions such as:-

      • What are the Per Codec Call Counts for Phone Groups?

      • What is the time based pattern for terminated calls (incoming and outgoing) for phones in the Phone Groups?

      • What is the time based pattern for the duration of the calls made by the phones in the Phone Groups?

      • What is the time based pattern for error reports associated with phones in the Phone Groups?

      • Which phone groups include phones with badly classified traffic that can damage VoIP QoS? - When & how often is this happening?

    • Reports covering when you have non VoIP traffic on your Voice VLAN.

 

Benefits when using Call Analyzer and Call Log -

  • Easy of Use

    • Just select the Phone, the rest is automatic, no need to set any kind of filters to view Phone and Server dialogs.

    • Has a simple interface with the ability to easily focus the display on just the important events in the Phone/Server dialogs.

  • Automatic tracking of calls.

  • You can use information provided to improve Call efficiency.

    • Can easily see when calls would have failed in the QoS engineering system, i.e., Calls that were not properly prioritized.

  • Long term independent Call Logging.

    • A SQL Call Log can be maintained to provide long term permanent call records.

    • The Toolbox has a simple yet powerful interface to filter/view call information in the Log. You do not need to be an SQL expert to access information in the SQL Call Log.

     

Benefits when using Frame Flow Analysis tool

  • Off line analysis

    • You do not need to be on site to find problems, you can just arrange for a frame file to be provided using Wireshark® or the Codima Protocol Analysis tool and apply the Frame Flow Analyzer to that file.

  • You can resolve QoS or Signaling problems on remote sites.

  • You can solve problems faster - Quicker analysis of frames files containing VoIP Traffic.

    • Can automatically break down contents of the frame file into dialogs – both incoming and outgoing. It then allows you to review the Signaling, Media and Reporting components.

  • You can identify equipment queues.

    • Analyses RTP media, letting you can see when the frames are bunched together.

  • You can quickly find out if frame drops are impacting on voice quality.

    • Analyses RTP media, showing frame drop patterns. Shows if frames are being dropped on a regular basis or being dropped in blocks. A few dropped frames on a regular basis may not cause a problem but being dropped in blocks will.

  • You can identify Load sharing problems, quickly find out if multiple paths are present.

    • Analyses RTP media, shows sequence error patterns. This type of problem can then be fixed by controlling load sharing setting on multi-link routes.

  • You can listen to Calls – hear what the voice quality sounds like.

    • Replays the RTP Media through a Media Player
       

For more information review the Features and Benefits Video Tutorial for the VoIP Monitoring and Troubleshooting Toolbox:-

Click here to access the sign in page for the Tutorials:-

 

or click here to obtain a copy of the VoIP Monitoring & Troubleshooting Toolbox Features and Benefits list.

 

 

 

What tools/functions are accessible via the autoVoIP tree branch?

The tools accessible via the autoVoIP tree branch include :-

  • VoIP Monitoring tool - provides monitoring functions for VoIP (SIP and Skinny) networks
  • VoIP Troubleshooting tool - provides Troubleshooting functions for VoIP (SIP and Skinny) networks

The functions accessible via the autoVoIP tree branch:-

  •   Provide a passive non-intrusive based VoIP analysis system
  •   Can service multi-site installations via locally installed probes
  • •  Can present enterprise network overview for NOC

Extra VoIP functions are also available via the autoAnalyzer tree branch, when you load a frame file, they cover :-.

  • Flow Analyzer - Facility to analyze the frame flow - breaking it down by Class, i.e., Media, Signaling and Reporting

  • RTP Analyzer - Facility to analysis the RTP payload

 

 

When would I use functions accessible via the autoVoIP tree branch?

  • During deployment of VoIP
  •  To check VoIP traffic service level - (SLA History graphs are provided)
  •  To monitor Call set up
  •  To Troubleshoot – i.e., establish if the network is suitable for VoIP
  • During daily operation
    • To monitor VoIP - Voice Quality is effected by other traffic so needs to be continually monitored
    • To Troubleshoot – i.e., quickly isolate and resolved problems experienced by users
    • To provide remote access to VoIP analysis

 

Scope and Operations

 

- Tracking RTP (Media)

Can I track RTP statistics for Skinny and SIP networks automatically without needing to provide RTP port information?

Yes, you can use the Traffic Police dashboard and the Phone Groups Manager dashboard.

Cisco Skinny - RTP port information is provided in the Skinny frames monitored by the Toolbox, so the system will automatically be able to track the statistics for RTP on the network being monitored.

SIP - RTP port information is provided in the SDP frames monitored by the Toolbox, so the system will automatically be able to track the statistics for RTP on the network being monitored.

The Toolbox must be correctly deployed to see the relevant traffic flow.

Example display - Showing Traffic Police dashboard - VoIP Frames per Sec fields and VoIP Bandwidth Bps statistics are shown as Cumulative Values and for the last 15 seconds.

Example display - when you select to drill down from the Traffic Police dashboard (Click on VoIP Frames per Sec fields to access)

Example display - Showing Phone Groups Manager dashboard - RTP All Phones field provides a overview and an access point to the History Charts covering, Codec usage, QoS, % Loss frames and Jitter for all the phones.

 

 

- Tracking Phones and Registrations :-

 

Can I control the Register Timeout, so system shows me the dead phones?

 

Yes, the register timeout used by the Toolbox is configurable, using a SIP Setting script.

The Troubleshooting Dashboard and the Phone Status Summary Dashboard (accessible via autoVoIP tree branch) show colored phone icons to denote the status of the phone, Dead Phones/Unregistered phones are shown as Grey. The Register time out is used to set phone to "Grey" status if the phone does not register within the defined period.

Excerpt from Troubleshooting Dashboard - showing unregistered phones in Grey:-

It is important when setting the timeout period that you make allowances for the register time associated with the phones on your network.

Example - If the Network Administrator sets the phones to all register at least every 5 Minutes and SIP Settings script has a Register timeout of 10 minutes, then the phone will go grey after 10 minutes of inactivity (not registering).

You can also use the Registrations Analysis Report to track down registration problems, see FAQ

 

Can I customize the Phone icon usage and the error statistics? - SIP only

i.e.

  • allocate error icons only for specified SIP response codes during specific request methods - INVITE or BYE for example
  • exclude/include specific SIP response codes from SIP Event and Error statistics

Yes, there are customization scripts for this purpose:-

  • SIP Icon organizer
  • SIP Error Code Actions

These scripts are accessed via the autoVoIP/Configuration/SIP/System branch of the Pane 1 Tree view.

Excerpt from SIP Icon Organizer script:-

 

 

Can I identify VoIP Servers on the network and independently track Registrations, see who is registering with them and find problems such as :-

  • Illegal phones attempting to access the VoIP Servers
  • Phones that are not getting replies and subsequently will have no dial tone
  • Excessive Registration Requests

Yes, the VoIP Server Analysis Dashboard lists the VoIP Servers monitored on the networks and you can drill down from there to the Server Phone Registrations Analysis dashboard to view the associated Registration information.

 

 

Can I monitor wireless phones?

The following url explains the situation in relation to monitoring wireless phones:-

To summarize, the Toolbox operating with an non wireless card could be deployed on the Internet side and monitor traffic in the normal way via a mirror port.
 

Can I see if phones are registering with more than one VoIP Server?

Yes, that information is provided in the Phone Registration List, accessible after you select a Phone in the Troubleshooting Dashboard or the Registrations Analysis Dashboard. See Phone Registration list example below

 

Can I see phone registration History?

Yes, that information is provided in the Phone Registration List, accessible from the Troubleshooting Dashboard and the Registrations Analysis Dashboard.

Registrations List example

 

Can Toolbox see phones that have not registered with the Registration Server?

The Phones have to be registered at some point to be properly monitored by the Toolbox.

There are two facilities that can be used to force phone display:-

  1. Load saved phones from an earlier session - this ensures that you do not need to wait for the Phone to re-register to use VoIP Monitoring facilities.
  2. SIP Only : Manually add the SIP Server IP address - this ensures that you can monitor traffic to/from the phone in situations where no registrations are being monitored.

Unregistered phones will be shown as grey icons in the Troubleshooting dashboard.

SIP error reports 606 for example will indicate that the Registration process is not being accepted.

You also have facilities to undertake an independent analysis of the Registration Process, both from the perspective of the VoIP Server and from the Phone, see below:-.

 

Can Toolbox track Phones in situations where the traffic has already passed through a NAT firewall and no longer contains the IP addresses of the Phones?

Yes - see FAQ entry :-

 

How are phones Tracked?

To track phones the Toolbox needs to use a unique ID, the options available are:-
  • By IP Address
  • By URI

The URI method for example would apply in a situation where the traffic being monitored by the Toolbox has already passed through a NAT firewall and no longer contains the IP addresses of the Phones. The tracking method used is controlled by the Toolbox License.

On a SIP network for example at least one phone has to have been monitored by Toolbox registering with the SIP Server to activate the phone tracking process. If the Toolbox is not seeing registrations it is also possible to by pass this stage by manually adding the SIP Servers IP address, see Help entry titled:-

  • How to manually add Registration Server

 

What has to be monitored for Toolbox to start tracking VoIP phones?

Phones are identified by monitoring the Registration process.

If you are monitoring a VoIP network and the Troubleshooting Dashboard is not showing the phones. You should check that :-

1. The Toolbox is properly deployed to capture the VoIP traffic - The Traffic Police Dashboard can be used to see the breakdown of VoIP Traffic being monitored.

See :-

2. The Phones are registering with the Registration server.

Note: There are also two facilities that can be used to force phone display:-

  1. Load saved phones from an earlier session - this ensures that you do not need to wait for the Phone to re-register to use VoIP Monitoring facilities.
  2. SIP ONLY : Manually add the SIP Server IP address - this ensures that you can monitor traffic to/from the phone in situations where no registrations are being monitored.

3. Provide Codima Support Department with a frame file captured using the Codima Toolbox or Wireshark™ that covering the phones SIP or Cisco Skinny (SCCP) traffic. The file can be analyzed to check the protocol dialogs and establish if there are Registration issues or if any special customization is needed.

See FAQ entry - If I have problems with Toolbox what evidence do I need to supply?

 

Will phone tracking work if my phones/SIP Servers do not use the Registered SIP Port (5060)?

The Toolbox can track SIP Traffic. The following dashboards are populated with information obtained by the analysis of the SIP protocol:-

  • Phone Analysis
  • Traffic Police
  • Trouble Shooting
  • Response Code Analysis
  • Registration Analysis
  • Phone Software Analysis

To do this it is important that the SIP Port is known to the system.

Registered Port

The Registered Port for SIP is 5060, so the Toolbox defaults to analyze traffic on that port as SIP.

Customization

The Toolbox can also be customized to decode SIP on other ports. Port 5061 for example has been added using the customization facility.

 

- VoIP Troubleshooting :-

Can I find calls that have used the wrong priority?

Yes, the Call Log will automatically highlight any calls that had the wrong priority.

 

Can I identify if specific phone types/software versions have problems?

Yes, you can use the Phone Software Analysis Report to view patterns effecting SIP phone types and software versions, e.g.,

The above example clearly shows the Phone types being effected by the different SIP Errors.   

 

Can I immediately spot problem phones?

Yes - using the Troubleshooting Dashboard, for example SIP Phones that have sent error reports will have Red, Orange or Purple phone status icons (subject to the error definitions in the Match Error Codes to Phone Icons configuration script). Clicking on the Error Phones Tab in this report will filter the Troubleshooting view to show only error phones.

Excerpt from Troubleshooting Report - showing error phones:-

You can use the find feature to locate Phone with specific types of error reports., e.g.,

You can also drill down and view error history, error information is provided chronological and by error type, e.g.,

 

Can I immediately spot registration problems and identify if they are solid or intermittent?

Yes - To establish if the problems are solid or intermittent, you can refer to the Phone Registration List, see

For more information on isolating Registration problems, see FAQs

 

- SLAs/QoS :-

Can I monitor SLAs?

Yes you can, a number of the history charts monitor SLA parameters, see example below.  For detailed information on this see Help entry titled How to check SLA.

You can also adjust the SLA bands associated with the display by using the Toolbox configuration facility.

 

Can I monitor QoS?

Yes, you can can monitor QoS :-

 

Can I monitor QoS for individual Phones?

Yes using Troubleshooting Dashboard

 

Can I monitor Global QoS?

Yes using Phone Groups Manager Dashboard

 

Can I monitor Calls, QoS and Codec usage for a user defined group of Phones?

 

Yes - using the Phone Groups Manager Dashboard, you can define different groups of phones based or IP address ranges or URL text strings. QoS, Call Counts and Codec Usage are tracked from the Groups, and the user can cycle through them in the Troubleshooting Dashboard display. This provides a method of focusing on a set range of phones.

 

 

 

When should Toolbox show QoS information - R and MOS values?

The Toolbox will calculate QoS if:-

  1. It has monitored QoS information in RTCP packets

Note: Just because you have RTCP frames being monitored by Toolbox, it does not mean they contain QoS information, for example they may just be SDES or BYE RTCP frames, which do not contain QoS.

The RR (Receiver Report) is used to primarily report QoS using the Fraction lost field.

Delay is measured using the DLSR (Delay since last Sender Report) field in the Receiver Report. So both RR and SR frames are needed to measure delay. The fields in the Receiver Report must contain valid information, i.e., not zeros or incorrectly formatted timestamps. Sender Report frames on their own will not provide QoS information. If both valid SR and RR frames are monitored the "R" values are more accurate as "Delay" is also used to determine QoS.

  1. It has also monitored Codec information.
  2. The Codec information is recognized - i.e., matches the list currently configured in the Toolbox.

                   Note: The Codec support can be enhanced using the Toolbox Codec Settings script.

If there is no suitable RTCP packets monitored to measure QoS, it will undertake the QoS calculation using RTP statistics to determine %  packets lost.

QoS is always calculated from the % dropped packets, the delay, and the Codec using formula ITU-T G.107 (03/2005) the E Model.

Jitter is also calculated using a standard formula (RFC 3550 page 33), the Jitter calculation is not used in the QoS 'R' value calculation.

Correct deployment of Toolbox is important if you wish to obtain RTP/RTCP frames as well as Signalling (SIP Traffic/Skinny Traffic), see Codima Toolbox Deployment Guide

Some protocols provide better QoS information than others, for example RTCP is better than RTP.

See FAQ entries titled:-

 

 

 

- Miscellaneous :-

Can I store phone lists and re-use next time I use the Toolbox?

Yes, there is a configuration option to save the monitored phone list and reload it when the Toolbox is restarted. The range of phones reloaded can be all the phones in the list or just the ones that were recently active (i.e., active within the last 24 hours). This facility allows you to access History Charts for Phones that are not currently active.

Does the Toolbox support secured SIP? - (SRTP)

This is not part of the standard service. If you have a requirement to monitor secured SIP (SRTP) you should contact Codima for information. To provide facilities to monitor secured SIP a Key would be required.

What are the issues associated with monitoring SIP Phones on Enterprise networks as opposed to at a SIP ISP?

Phone communication and SIP protocol support:-

In an Enterprise environment SIP phones typically communicate internally within the campus and also over WAN/MPLS routes. They typically use standard hardware based Phones, which follow the protocol standards.

A SIP ISP sits in the middle of a lots of consumer/SMI type subscribers, who will be using all types of phones. They will typically use Soft Phones, which do not always follow the protocol standards exactly.

The Toolbox has been customized to support some variations of the protocol standards, but users should be aware of that this is a potential danger area.

Tracking Phones

You also need to consider what method needs to be used to track the phones. To track phones the Toolbox needs to use a unique ID, the options available are:-

  • By IP Address
  • By URI

The URI method for example would apply in a situation where the traffic being monitored by the Toolbox has already passed through a NAT firewall and no longer contains the IP addresses of the Phones. The tracking method used is controlled by the Toolbox License. See FAQ:-

Measuring QoS :-

Another issue is how to measure QoS for the SIP ISP. Because the SIP ISP sits in the middle of Incoming and Outgoing WAN Links, RTP QoS is only ever measured half way. The only way End-User QoS can be determined is by using RTCP.  In the ISP environment, you could potentially make us of Probes on the customer site to undertake QoS measurements when the RTP traffic is not available to the ISP.

Its different with the Enterprises environment as QoS can be measured at the network edge, so if phones do not support RTCP you can still measure QoS using RTP, however even in an Enterprise environment you get better QoS information from RTCP.

Other issues :-

In an Enterprise environment you can also make use of the functions accessible via the autoVoIP Simulator tree branch to Monitor QoS continuous between the Enterprise sites.

 

What do the different Phone Icons mean?

The Troubleshooting Dashboard and the Phone Status Summary Dashboard show colored phone icons to denote the status of the phone.

The Troubleshooting Dashboard applies this to both the current state and past state, the past state icons are generally smaller than the current status icon.

Display showing current and past state icons:-

- Error reports currently and in the past

. - Good calls now, error reports in the past

- Good calls now, activity transmitting

- Phone Registered

Phone Icon type allocation

The status/error codes assigned to SIP phone icon types is configurable.

Phone Colors

  • Grey =Trying to Register or Dead Phones/Not Registered - dead phone status is controlled by the configured Register Timeout
  • Blue - Registered
  • Green - Good Calls
  • Good Call Status icons : Green Icon with added display icon covering status, e.g., Parking

Controlled by the code definitions in the Match Error Codes to Phone Icons configuration script.

  • Red, Orange, Purple, Brown, Pink - Errors
  • Important errors : Red icons, with added display icons covering type of error, e.g., No Access
  • Non Critical errors : Orange, Purple, Brown and Pink icons - SIP only

These are subject to the error definitions in the Match Error Codes to Phone Icons configuration script.

 

 

- Performance/Network type :-

Can I use Toolbox on SIP networks?

Yes

Can I use Toolbox on Skinny networks?

Yes

Can I use Toolbox on H323 networks?/Can I use the functions accessible via the autoVoIP tree branch on any VoIP network?

The functions accessible via the autoVoIP tree branch are specifically designed for use on SIP or Skinny networks.

However if you an environment that includes for example a SIP - H323 Converter, see diagram below, then you can use the Toolbox to monitoring the SIP signaling call setup/clearance, but not the H323 call set up/clearance.

However Codima does supply the VoIP Simulation tool and the VoIP Pre Deployment Assessment tool. These are tools that operate solely using RTP/RTCP packets and as such can be used on any VoIP network to measure QOS. These functions are accessible via the autoVoIP Simulator and autoVoIP Blaster tree branches.

The VoIP Simulation tool is part of the VoIP Monitoring & Troubleshooting Toolbox and the VoIP Pre Deployment Assessment tool is part of the VoIP Readiness Toolbox.

 

How fast can Toolbox handle traffic/what line speeds will it support?
 
The line speed is handled by the NIC card used, i.e., if it is a 10Mb NIC it will monitor traffic at that rate, if it is 1Gb NIC it will monitor traffic at that rate.

 

The performance will be subject to the capabilities of the NICs used by the Toolbox, dropped packet counts are logged.

 

For additional performance information, see entry titled:-

- Phone/Codec Support :-

Does Toolbox support header compression, VAD (silence suppression) and a variety of Codecs?

The Toolbox does not currently do header compression as there has not been a large demand by our client base.

VAD is not part of the standard Mos, R value formula.

The Toolbox does support a number of Codecs and can it is possible to add additional Codecs to the system. This is an expert user function and is done using a Codec Configuration script, which will requires the Ie and Bpl values for the Codec. Ie identifies the quality degradation due to the decoder and Bpl the robustness of the PLC algorithm against packet loss.

 

What Codec support is provided?

The Toolbox supports a number of Codecs and can add additional Codecs to the system. This is an expert user function and is done using a Codec Configuration script, which will requires the Ie and Bpl values for the Codec. Ie identifies the quality degradation due to the decoder and Bpl the robustness of the PLC algorithm against packet loss.

The current Codec support includes:-

See below for more information on Voice Codecs

Voice Codec

Codec is a piece of computer hardware or software used for the compression and/or decompression of digital media (most usually audio or video). QoS may be modeled mathematically from the Codec.

Here is a list of some of the voice codecs standards:-

Codec Algorithm Bit rate (K bits per second) Notes
G.711 PCM 64, 56 A ITU (International Telecommunications Union) standard for a narrow-band audio codec that encodes speech into a stream of 8 bit samples (or less frequently 7 bit samples) at 8khz. This creates a data stream at either 64kbps or 56kbps. G.711 uses a logarithmic mapping that emphasizes the parts of the signal that the human ear is most sensitive to. Uses pulse code modulation

There are two variants of G.711-

  • uLaw - Used with T1 and J1 connections
  • aLaw - used with E1 connections (Europe and Australia)

High quality, high bandwidth
G.722 ADPCM 48-64 A ITU standard wideband speech codec.

G.722.1 offers low bit-rate compressions.

G.722.2, also known as AMR-WB (Adaptive Multirate Wideband), offers even lower bit-rate compressions and the the ability to vary compressions as the network topography changes. For example bandwidth is automatically conserved when network congestion is high. When congestion returns to a normal level, a lower-compression, higher-quality bit rate is restored.

G.722 and its variants sample audio data at a rate of 16 kHz, double that of traditional telephony interfaces, which results in superior audio quality and clarity.

G.722 patents have expired, so it is freely available
G.723.1 ACELP 5.3 A ITU standard for a narrow-band audio codec that encodes speech into a stream of data frames that each represents 30ms (240 samples) of speech data. Each frame can be either 24 or 20 bytes long, which makes the data stream either 6.4kbps or 5.3kbps.

License fee to use this Codec commercially. 

Lower quality and < 10% of bandwidth used by G.711

(Less bandwidth = more delay)

Uses MP-MLQ (Multi-Pulse Maximum Likelihood Quantization)
G.723.1 MP-MLQ 6.3
G.726 ADPCM 40, 32, 24, 16 A ITU standard for a narrow-band audio codec that encodes speech into a stream of 2, 3, 4, or 5 bit samples - data stream = 16kbps, 24kbps, 32kbps, or 40kbps. 

Uses ADPCM (Adaptive differential pulse code modulation.
G.728 LD-CELP 16 A ITU standard for a narrow-band audio codec that encodes speech into a stream of 10 bit frames that each represent 5 samples - data stream  = 16kbps. License fee to use this Codec commercially. 

Uses LD-CELP ( Low-delay code excited linear prediction)
G.729 CS-ACELP 8 A ITU standard for a narrow-band audio codec that encodes speech into a stream of data frames that each represent 10ms (80 samples) of speech data. Each frame is 10 bytes - data stream = 8kbps. License fee to use this Codec commercially. 

Uses CS-ACELP (Conjugate-structure algebraic-code-excited linear prediction speech coder)
GSM 06.10 PCM 13.2

A narrow-band audio codec that encodes speech into a stream of data frames that each represent 20ms (160 samples) of speech data. Each frame is 264 bits, giving a data stream of 13.2kbps.

 

What VoIP phones are supported? What soft phones are supported?

Codima does not publish a list of supported phones as the Toolbox is designed to operate by monitoring traffic that either :-

Some customization (which will potentially impact over a range of soft phones for example) has been undertaken to handle some variations in the SIP protocol implementation. If you have issues with phones, a frame file containing a typical call set up and clearance for the call will enable Codima to establish if customization is relevant.

There is also a user configuration facility that can be used to handle some protocol variations, for example switch off or on the option to use Session Description Protocol information in the "180" SIP response code when calculating QoS.

 

- Call Analysis/VoIP Call Group Analysis : Statistics gathered by Group (GAG)

What level of Call analysis can be undertaken?

Call Analysis using Frame Decode facilities

The Toolbox includes a Call Analyzer facility that can be used to decode the Signalling and Reporting components of a call.

VoIP Call Group Analysis : Statistics gathered by Group (CAG)

The Toolbox can perform in depth analysis on completed VoIP calls and provide statistics covering :-

This is undertaken using the Phone Groups Manager Dashboard.

 

 

 

- Protocols :-

What VoIP protocols does Toolbox support? - What type of VoIP Networks can it be used on?

This will depend on the tool being used :-

The VoIP Monitoring tool and the Troubleshooting tool (accessible via autoVoIP Tree branch) : part of the VoIP Monitoring & Troubleshooting Toolbox and the Codima All in One Toolbox.

Can be used on VoIP networks that use Skinny or SIP Phones, it analyses information held in SIP, SDP, Skinny, RTP and RTCP frames:-

  • SIP :  Session Initiation Protocol = Protocol associated with Call set up and clearance
  • SDP : Session Description Protocol = Protocol used to describe the details of the streaming media sessions and multicast transmissions
  • RTP : Real Time Transport Protocol = Protocol used for the transmission of video and audio files in real time for Internet applications such as IP telephony.
  • RTCP : RTP Control Protocol (RTCP) = Protocol associated with Quality of Service (Q0S)
  • RTCP-XR : RTP Control Protocol, Extended Reporting = Protocol associated with Quality of Service (Q0S)
  • Skinny (SCCP) : Skinny Call Control Protocol (SCCP) is a proprietary network terminal control protocol owned and defined by Cisco Systems Inc as a messaging system between a Skinny client (VoIP Phone) and the Cisco Call Manager.

H323 is not currently supported.

Mirror ports should be set up to capture only one type of VoIP traffic during an  session, either SIP with RTP/RTCP or Skinny with RTP/RTCP.

The VoIP Simulation tool (accessible via autoVoIP Simulator tree branch) : part of the VoIP Monitoring & Troubleshooting Toolbox and the Codima All in One Toolbox.

Can be used on any VoIP network as it uses only RTP and RTCP frames, there is no requirement to monitor Call Set up or clearance. The Tool measure QoS using information in RTCP frames.

The VoIP Pre Deployment Assessment tool (accessible via autoVoIP Blaster tree branch) : part of the VoIP Readiness Toolbox.

Can be used on any VoIP network as it uses only RTP and RTCP frames, there is no requirement to monitor Call Set up or clearance. The Tool measure QoS using information in RTCP frames.

 

- Flow Analyzer :-

 

Can I load and analyze a captured trace file covering VoIP traffic? - Post Capture frame file analysis

It is possible to replay frame files captured using the Passive Analysis tool included in all the Codima Toolboxes, as well as trace files from Wireshark™

This is done using the Flow Analyzer, which breaks down the calls to/from SIP phones into dialogs, covers Signaling, Media and Reporting for both the incoming and outgoing flow.

 

The frame files should include RTP frames to make the most of the Flow Analyzer, as the RTP payload can be analyzed further using the following specialist facilities:-

 

Note: Frame files capture via the Toolbox Call Analyzer are NOT suitable for RTP Analysis as they do not include the RTP payload.

 

 

What is the Flow Analyzer  - what do I need to access it, what does it provide?

 

The Flow Analyzer is accessible via the autoAnalyzer tree branch, when a frame file is loaded. The Flow Analyzer breaks down the calls make by SIP phones into dialogs, covering Signaling, Media and Reporting for both the incoming and outgoing flow.

 

The Flow Analyzer operates post capture on frame files captured using the following tools:-

 

The frame files should include RTP frames to make the most of the Flow Analyzer, as the RTP payload can be analyzed further using the following specialist facilities:-

 

Note: Frame files capture via the Toolbox Call Analyzer are NOT suitable for RTP Analysis as they do not include the RTP payload.

 

 

 

 

 

 

- Call Playback Facility :-

 

The Call Playback facility replays calls using a media player, see below:-.

 

Can I listen to calls?

 

Yes - you can use the Call Playback facility available via the Flow Analyzer. The Call Playback can only be accessed by users with a suitable License.

 

The facility is accessible via the Frame Flow Analyzer.  It operates post capture and can be applied to frame files captured using the Passive Analysis tool included in all the Codima Toolboxes as well as trace files from Ethereal™/Wireshark™. The frame files you use with this feature must include RTP frames. Frame files captured via the Toolbox Call Analyzer are NOT suitable as they do not include the RTP payload.

 

To access this facility you with load a frame file (using functions accessible via the autoAnalyzer tree branch) and have a suitable media player installed on your Host Platform. 

- RTP Analyzer :-

The RTP Analyzer is a specialist facility to analyze the RTP payload, see below:-

Can I analysis the RTP Payload?

- e.g.,

 

Yes, the RTP Analyzer is used for this purpose. The RTP Analyzer is a specialist facility to analyze the RTP payload. The facility is accessible via the Frame Flow Analyzer. It operates post capture and can be applied to frame files captured using the Passive Analysis tool (provided in all the Codima Toolboxes) as well as trace files from Ethereal™/Wireshark™. The frame files you use with this feature must include RTP frames. Frame files captured via the Toolbox Call Analyzer are NOT suitable as they do not include the RTP payload.


Display showing access points from Flow Analyzer to all drill down options, including RTP Analyzer.

 

 

 

 

 

Example showing the RTP Analyzer interface:-

 

 

Can I use the Flow Analyzer with Frame files in the Call Analyzer Frames File folder? 

Can I use the Call Playback with Frame files in the Call Analyzer Frames File folder? 

Can I use the RTP Analyzer with Frame files in the Call Analyzer Frames File folder? 

No. Frame files captured via the Toolbox Call Analyzer are NOT suitable for the Flow Analyzer, RTP Analyzer or Call Playback as they do not include the RTP payload.  

The Flow Analyzer, RTP Analyzer and Call Playback are applied to frame files captured using the Passive Analysis tool included in all the Codima Toolboxes as well as trace files from Ethereal™/Wireshark™.

 

- Reports

 

What are the benefits of using the Reports Manager? - when using the VoIP Monitoring and VoIP Troubleshooting tools

 

The key benefit is it that this tool provides you with evidence at your finger tips. Producing a wide range of Reports that can be used to show both network trends and isolate problems. The VoIP Monitoring and VoIP Troubleshooting tools include:-

  • Reports covering the history of phone group traffic patterns gathered using passive analysis. They will enable you answer questions such as:-

    What are the Per Codec Call Counts for Phone Groups?
    What is the time based pattern for terminated calls (incoming and outgoing) for phones in the Phone Groups?
    What is the time based pattern for the duration of the calls made by the phones in the Phone Groups?
    What is the time based pattern for error reports associated with phones in the Phone Groups?
    Which phone groups include phones with badly classified traffic that can damage VoIP QoS? - When & how often is this happening?

  • Reports covering the history of VoIP traffic patterns gathered. This would include a report covering when you have non VoIP traffic on your Voice VLAN.

 

For more on the Reports Manager tool - see FAQs - Reports Manager

 

For more on Features and Benefits or click here to obtain a copy of the VoIP Monitoring & Troubleshooting Toolbox Features and Benefits list.

 

 

What kind of reports are available?

 

There are a number of different types of Reports, they include :-

  • HTML Reports created using information in History Charts and Traffic Police display.

  • Pre defined Statistical Reports - accessible via the Reports Manager tree branch, these reports can be produced as HTML Reports or scheduled as daily, weekly, monthly Word reports.

 

For additional information, see FAQ - Reports Manager

 

 

- SQL Call logging

Can I have some guidance covering how to install, configure and use the SQL Call Logging tool?

Yes, the "Operations Guide for the autoVoIP Tree branch and other linked functions" has a section on the SQL Call Logging tool. It covers the 4 stages required to operate this tool :-

  1. Installing MySQL Server
  2. Configuring system to write to SQL Database
  3. Accessing SQL Call logging search facility – Supplying search parameters
  4. Viewing results

 

Information is also provided in the online help accessible via <F1> - Help entry titled "SQL Call Logging"

 

There is also a short Tutorial titled

  • How to install, configure and use the SQL Call Logging tool

Click here to access the sign in page for the Tutorials:-

 

 

Can I store call information in a SQL database?  Can I log Call information for an extended time period?

Yes, this is available for SIP and Cisco Skinny networks, using the SQL Call Logging tool.

The SQL Call Logging tool is an Add on for the Codima VoIP Monitoring & Troubleshooting Toolbox and the Codima All in One Toolbox. The user can select to write call information to a SQL Database. The database can then be browsed/filtered by the Toolbox and the results presented in the Call Log, where the user can also access the Call Analyzer, which allows the user to view Phone/Server dialogs associated with the selected Call.

The filter can be applied to:- 

  • A selected time period

  • All phones or to/from listed Phone(s)

  • Good Calls/Bad Calls/Calls with specific errors

  • QoS values and bands

  • A combination of the above

 

Display showing the process used:-

 

Display showing SQL Database excerpts:- 

 

  

Advantages :-

  • The user does not need knowledge of any SQL language to use the SQL Call Logging tool

  • The call logs from multiple Toolboxes can be sent to the same SQL server to aggregate calls from multiple sites.

  • The SQL server can be local or remote to the Toolbox platform logging the calls.

  • The Call information is stored in a SQL Database, which allows the user to write their own SQL queries and make use of standard SQL report generator packages to :-

    • View and organized the call information

    • Convert from SQL to XML and CSV formats

 

For latest information see  - "Operations Guide for autoVoIP Tree branch and other linked functions including the autoVoIP SQL Call Logging Tree branch" provided with the SQL Call Logging Add on

 

Do I need to be able to write SQL queries to use the SQL Call Logging tool?

No, the user does not need knowledge of any SQL language to use the SQL Call Logging tool, SQL Queries are automatically created from the user provided search parameters.

 

 

- Alarm Reporting/Trap issue

 

What are the Benefits of using the Global Alarm System?

  • You can be proactive : Automatic notification of threshold breaches.

  • You have multiple alarm reporting methods.

    • Ensuring that you can integrate the tool with your operating practices, for example if you have an SNMP Management system, the alarms can be reported to it as SNMP Traps or if your engineers prefer email notifications or SMS text messages, then alarms can be reported that way.

  • You have controlled Alarm Reporting

    • Flood control ensures you are not overloaded with alarm reports - can avoid generating an alarm report for very short lived events and stop the same alarm condition being logged too many times.

    • Email and SMS Alarm report frequency is controlled - multiple alarms reports are included in single Emails/SMS Messages.

  • You have predefined alarm thresholds on key events - ensuring system works out of the box.

  • You have an embedded email client – independent of the Networks own email facilities, so will be able to report on any failures associated with the platform hosting the Networks email client.

 

For more information on Features and Benefits click here to obtain a copy of the VoIP Monitoring & Troubleshooting Toolbox Features and Benefits list.

 


 

 

Can I have alarm reports automatically emailed to me?

Can I set SNMP Traps?

Can I set threshold alarms?

 

Yes. All the Codima Toolboxes use a Global Alarm System to log and report Threshold alarms. The Global Alarm System can be set up to apply one or more of the following actions

• Log the alarm report
• Send out an SNMP Trap when a threshold value is breached (goes above threshold setting) or when a value drops (goes below threshold setting).
• Send a notification email or SMS text message when a threshold value is breached (goes above threshold setting) or when a value drops (goes below threshold setting).
 

For detailed information on this facility, see Help entries titled

  • How to email alarm reports

  • How to set SNMP Traps

  • How to set alarm thresholds

  • How to configure Email Client

 

  

Can I integrate with SNMP Managers?

The Codima Toolbox includes tools which provide SNMP Manager Integration.

This is done by accessing the fully integrated SNMP Module. This module can add value to an already installed SNMP Manager, in a number of ways. Including the following:-

EXTEND MANAGERS RANGE TO COVER NON-SNMP NODES/TRAFFIC

  • Issue Traps to the Manager in respect of problems associated with Non-SNMP Compliant Nodes, immediately extending the SNMP Managers range.
  • Issue Traps in respect of all the Traffic/Errors on a Segment. (i.e., monitoring actual real time loading and errors, not the loading/error information obtained from SNMP compliant nodes), providing the manager with a complete/impartial view of the segments traffic.

This will use information obtained from Passive Analysis of the Network.

 

Traps can be issued to multiple SNMP Managements systems, for full information, see Help entry titled:- How to set SNMP Traps

 

The SNMP Manager must have compiled the Codima MIB. The MIB {Enterprise.226} is included in the file set installed with the Codima Toolbox.

..:\Program Files\Codima\Express\SNMP\CODIMA-EXPRESS-MIBs\
 

- Cisco Call Manager

Does the Call Log provide more information on calls than Cisco Call Manager?

There are some advantages associated with the method the Toolbox uses to track calls when compared to Call Manager.
 
CDR or Call Detail records are produced by Call Manager, not by any Manage Engine Software, they just parasite it.
Call Detail records contain only basic details like parties and duration. There is no QoS or Performance information at all.
 
The Toolbox by contrast, tracks absolutely everything and also has the option of writing its own CDR call log records to an SQL Server. Which provides the user with long term call records that allow for multiple access. (SQL Call Logging tool).
 
The functions accessible via the autoVoIP tree branch, enable the Toolbox to analyze the SLA of Call QoS, Jitter and Frame Loss, they also provide access to the Call Analyzer, which maintains an abbreviated Graphical Call Analysis for every Call.
This facility is also available with the SQL Call Logging tool.

 

 

- Remote Operations:-

How would I monitor VoIP traffic on multiple networks? - Remote Operation

All the Codima Toolboxes include facilities to undertake remote management. A minimum of two Host Platforms are required:-

  • Platform 1 = Toolbox - Remote Manager
  • Platform 2 = Probe

The probes would operate independently and the manager would be able to:-

  • Open and view dashboards (accessible via the probes autoVoIP tree branch), for example view the Troubleshooting dashboard and Traffic Police dashboard, real time updates are provided.
  • Operate drill down features on probe, to access Call Logs, Call Analyzer, History views and reports and Automated Correlation Engine
  • Retrieve frame files from probes to allow for operation of the Frame Flow Analyzer, the Call Playback facility and the RTP Analyzer.
  • Open and changes configuration parameters on probe.

 

What is the Remote Manager - what is it used for?

The Remote facilities provide remote viewing and control. A Remote Manager is used to view and in some cases control the Remote systems, These remote systems can be:-

Type of Remote system What is it used for:-

 

 Enables you to use the functions accessible via remote systems autoAnalyzer and autoVoIP tree branches. Extends the domain being monitored by the Passive Analysis tool
  • Toolboxes on the remote site
    •
Enables you to use the functions accessible via the remote systems autoMonitor or autoPinger tree branches.
  • Sink systems designated as Blaster Managers

 

 Enables you to remotely control the stress testing process used by the VoIP Pre Deployment Assessment tool

For more information, see FAQ - autoVoIP Blaster Tree branch and the VoIP Pre Deployment Assessment tool section in the Codima Toolbox Deployment Guide

 

It provides the console view to show the information supplied by the Probes.

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What are Probes - what are they used for?

 

They are Remote Systems that provide the Remote Manager with access to the following tree branches on the remote system.

 

autoVoIP tree branch  - used by:-
  • VoIP Monitoring tool - provides monitoring functions for VoIP (SIP and Skinny) networks.
  • VoIP Troubleshooting tool - provides Troubleshooting functions for VoIP (SIP and Skinny) networks.

autoAnalyzer tree branch  - used by:-

  • Passive Analysis tool - provides statistical analysis, includes Live Views covering real time traffic patterns.

  • Protocol Analysis tool - provides real time packet analysis and an expert system to analysis captured frames.

 

They are used to provide local management, monitoring and analysis facilities. The following diagrams shows how multiple Probes can be deployed to extend the domain managed:-

 

Displays showing the Remote Manager and Probe deployment :-

 

 

 

 

 

- VoIP Technology

How do I find out what the SIP error codes mean?

Use the Toolbox help facility, access using <F1> key and search for "SIP Response Codes".

What is meant by QoS?

This term applies to a system used to provide different prioritization levels for different types of traffic over a network. Protocols used to achieve the required quality of service, including the RSVP, VLAN Priority, IP DiffServ.

For example, VoIP or streaming Video should have a higher priority than Mail traffic, as the consequences of interrupting Voice traffic or streaming Video are more obvious than slowing down Mail traffic.

Four Key parameters to determine QoS are:-

 

QoS is subjective as measured by experimentation with human listeners - SUBJECTIVE QoS

QoS may be modeled mathematically from Codec, %loss/delay to predict - OBJECTIVE QoS

These two should be similar –That is the challenge of the Mathematical Model to predict subjective QoS accurately.
 
Codecs cope with Jitter by using a Time Delay Q called a Jitter Buffer. This imposes a delay but allows Jitter up to this maximum to be ironed out. Packets outside the delay are counted as LOST - Long QoS. Jitter buffers eat into the delay budget but are more robust to jitter (and subsequent packet loss). So recently Dynamic Jitter buffers are being used to auto adjust to actual Jitter and minimize delay.
 
Delay comes from the Codec, it is substantial for compressing Codecs like 723 (not 711) and serialising Voice Samples into a RTP Packet. Network Queues in routers (usually at Edge Router), Distance due to speed of Light/Electricity in cables.
 
So lots of interdependent things eat into the Delay Budget. Using a High Compression Codecs saves HUGE bandwidth but makes system more fragile and Network QoS dependent.

 See also FAQ entries titled:-

 

§What are MOS and R values?

– Shown in the QoS entries in the Troubleshooting Report for example.
§
§

MOS (mean opinion score : standard - P80) is a Metric intended to convey User Experience of Phone Conversation in a single number

§

R-value (standard – G107) is an objective measurement calculated directly from measurements of packet loss, jitter and delay. It also has a strong correlation with the MOS value.

What is RTCP-XR?

RTP Control Protocol Extended Reports, this is a new VoIP management protocol, it defines a set of metrics that contain information for assessing VoIP call quality and diagnosing problems. See RFC 3611 for detailed information. This protocol is not yet widely used commercially.

RTCP-XR lets systems like the VoIP Monitoring tool obtain information on signal, noise and echo levels without having to decode the packet stream, which will be essential if payloads are encrypted using the new Secure RTP protocol from the IETF.

§
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Why is RTCP a more effective method of measuring QoS, than measuring Jitter/Lost Frames in RTP Directly

For the following reasons:-

§RTCP reports the user experience because it alone (in general) experiences the final jitter and lost frames values at the end point phone as experienced by the user. §Measuring RTP at a SPAN port is susceptible to frame drops and queuing delays caused by the SPAN mechanism itself, there by potentially effecting the QoS results.

 
In contrast RTCP, lost or delayed frames do not cause calculation errors, instead there are just blips in the QoS feed. There is a much smaller bandwidth required for monitoring RTCP, presenting a much lower load to the SPAN port.
 
The phones are distributed throughout the Network, and each phone would need to be monitored locally to get an accurate RTP QoS calculation
 
RTCP reports the user Experience at Both Ends of the RTP path, which is not possible to determine when monitoring RTP at a single point.

VoIP Traffic Flow

The VoIP Monitoring and VoIP Troubleshooting tools need to passively monitor the signaling (SIP, SDP frames or Skinny) and measure Quality of Service. QoS can be calculated from the information in RTCP frames issued by the phones or can be calculated by monitoring the RTP media (if phones do not support RTCP),  so it is critical that they are deployed correctly to ensure that they can see the relevant traffic. The ability to monitor Skinny, RTP and RTCP frames or SIP, RTP and RTCP frames from a single point is subject to a number of issues:-

  • How you network is structured
  • What mirror/span port capabilities your switches have
  • The Phone configuration, for example do the phones send RTCP information to the SIP server

§The Toolbox should typically be on the VoIP VLAN and if Skinny, RTP and RTCP frames or SIP, RTP and RTCP can not been seen from a central point you can use the multi-port facility to pipe both sets of traffic to the Toolbox. For guidance see :-

You may also need to consider using multiple Probes if you traffic is distributed, see FAQ titled:-

When working out how to deploy you should consider the traffic flow on your network, the following diagram show some typically traffic flows :-

SIP Networks

SIP and H323 Networks

Skinny (SCCP) Networks

- Automatic restart

 

Can I automatically reload the Toolbox when Platform is powered on?

 

Yes, the process is exactly the same as it would be for any other applications, i.e., you include a short cut to the application in your Start up folder.

 

This facility is especially relevant to Probes.

 

- Other Information

Managed devices

Managed devices are any device the Codima Discovery Engine get management information from, i.e., not ones that are just addresses in the ARP/Routing/Forwarding tables or devices that we just know exist because they respond to pings and reverse DNS, but devices that give the system information, such as the type of device it is or in the case of SNMP and WMI provide a lot of internal information on configuration etc.
 
The Discovery engine uses a number of methods including SNMP, WMI, CDP, SIP Queries and NetBIOS to get management information.

 

Support/Troubleshooting

Can I have a quick start guide - covering just the basic checks to get me started?

Yes - it is part of the Quick Start and Troubleshooting Guide.

Display showing the Quick Start Guide:-

Full Check List :-

Can I have a check list to cover what is needed to successfully operate the functions accessible via the autoVoIP tree branch?

 

Yes, click below to obtain Check list/Troubleshooting Guide:- 

 

 

How do I check that the Toolbox Host Platform is correctly deployed and configured to enable me to monitor traffic?

The Host Platform must be correctly deployed to be able to see the relevant Traffic Flow, e.g., attached to a suitably configured mirror port or tap.

The Host Platforms Network Adapter(s) must be correctly configured and connected to the network to monitor traffic. For guidance see :-

 

If I have problems using functions accessible via the autoVoIP tree branch what evidence do I need to supply?

click below to obtain a hard copy of the evidence requirements for the Toolbox.

 

- Frame Capture (to supply evidence)

If I want to capture frames covering just the VoIP traffic what do I need to do?

This would involve applying pre capture frame filters.

There are several approaches

  • You can set a filter based on a dialog, e.g., between a phone and its VoIP Registration server.

To capture this kind of dialog the filter set up would look like this:-

Example

  • Phone = 10.0.1.201
  • VoIP Sever = 10.0.1.200

  • You can set a filter covering traffic to/from the VoIP Registration server

To capture this traffic the filter set up would look like this:-

Example

  • VoIP Sever = 10.0.1.200

  • You can set filters for the Protocols. This is harder as the ports used in some cases are dynamic, so it is easy to miss important frames.

To capture SIP traffic typically the filter would look like this:-

To capture RTP at the same time, you can use the Filter 2 tab, the filter would look like this (if RTP is using UDP port 5004)

Capture Buffer showing both RTP and SIP Traffic

To use two Filter configurations in parallel as illustrated, you must set both Filter configurations to the Filter Active state, before capturing frames.

The tutorial titled "How to capture, filter and save frames" provides more information on using the frame filter facilities. The tutorial is accessible from the Help facility and from the Codima Website.

Click here to access the sign in page for the Tutorials:-

 

 

If I am using VoIP Monitoring and VoIP Troubleshooting tools on a busy network with lots of non VoIP traffic, can I improve performance by focusing processing on just VoIP frames?

Yes - you can use the VoIP Settings configuration options shown below, to focus on :-

  • VoIP only traffic (Signalling, Reporting and Media)
  • VoIP Signalling and RTCP only (Signalling and Reporting)

 

 

Microsoft® Patch level

What is the latest Microsoft® Patch level that the Codima Toolbox software been tested with?

The process of testing with Microsoft® patches is an ongoing one, latest level tested is as follows :-

  • Windows XP Service Pack 3

  • Windows 2003 Server SP 2

  • Windows Vista SP1

  • Window Server 2008 SP1

  • Windows 7

 

 
 

 

 

 

 Copyright/Disclaimer

Copyright ©2011 Codima Inc. All Rights Reserved.
No part of this document may be copied or reproduced in any form or by any means without the prior consent of Codima Inc.
Information in this document is subject to change without notice and does not represent a commitment on the part of Codima Inc. Codima Inc. reserves the right to revise or change this document without obligation of Codima Inc. to notify any person of the revision or changes. Information in this document is believed to be accurate at the time of publication but no liability whatsoever can be accepted by Codima Inc. arising out of the use of this information. Also, this document could include typographical errors or technical inaccuracies.
Any trademarks or trade names owned or registered by any other company and used in this document are the property of their respective companies

  

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