User Guide EXpert VoIP Test Tools
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Contents 1 Introducing the EXpert VoIP Test Tools ....................................................... 1 Conventions ............................................................................................................................2 2 Getting Started ............................................................................................ 3 Signal Connection ...................................................................................................................
Test Results ..................................................................................................53 Call Details ...........................................................................................................................54 Call Manager Statistics (H.248/Megaco) ..............................................................................56 Call Manager Statistics (H.323) ............................................................................................
1 Introducing the EXpert VoIP Test Tools The EXpert VoIP Test Tools provides Ethernet testing capability on the EXFO’s platform. It runs as a separate software application on the platform and operates independently of the modules installed on the platform. The EXpert VoIP Test Tools application generates a single VoIP call from the platform to another EXFO’s platform supporting the application or any IP phone. The call can either be "live", “stored audio” or “synthesized media”.
Introducing the EXpert VoIP Test Tools Conventions Conventions Before using the product described in this guide, you should understand the following conventions: WARNING Indicates a potentially hazardous situation which, if not avoided, could result in death or serious injury. Do not proceed unless you understand and meet the required conditions. CAUTION Indicates a potentially hazardous situation which, if not avoided, may result in minor or moderate injury.
2 Getting Started Signal Connection The EXpert VoIP Test Tools uses the 10/100/1000 Mbit/s Ethernet RJ45 port of the EXFO’s platform. Connect the 10/100/1000 Mbit/s electrical signal to be tested to the RJ45 port of the EXFO’s platform. Note: Refer to the EXFO’s platform User Guide for more information. Laser Safety Information The EXpert VoIP Test Tools software is not provided with any hardware components. However, it may be used with your platform or modules which may contain laser components.
Getting Started Starting the EXpert VoIP Test Tools Application To start the EXpert IP Test Tool application: 1. FTB-1: From Mini ToolBox, tap on the Test Tools tab. FTB-2 and FTB-2 Pro: From ToolBox X, tap the Test Tools button. FTB-200v2: From Compact ToolBox, tap on the Test Tools tab. 2. Select the EXpert VoIP Test Tools then tap on Start. At startup, the application always loads the previous configuration.
3 Using the Graphical User Interface This chapter describes the graphical user interface of the EXpert VoIP Test Tools application. Main Application Window The following main application window is displayed when the EXpert VoIP Test Tools application is started. Title Bar Test Control Main Window Main Menu Test Results Application Buttons Status Bar Main Window The main window allows to setup a test and to view the test status and results.
Using the Graphical User Interface Main Application Window Status Bar The status bar displays the following information. Icon and/or text Description Test name Test name of the selected test. Link arrow Green arrow: Ethernet Link up Red arrow: Ethernet Link down. Interface speed 10 Mbit/s, 100 Mbit/s, or 1000 Mbit/s. Test status Current test status. Title Bar The title bar displays the software application name and the battery level indicator.
Using the Graphical User Interface Main Application Window Application Buttons About (i) mainly displays the product version details and technical support information. Software Options button displays the list of software options enabled. For information on how to install and activate software options, refer to the EXFO’s platform User Guide.
Using the Graphical User Interface Arrow Buttons Arrow Buttons Moves to the top of the list. Moves one page up. Moves one line up. Moves one line down. Moves one page down. Moves to the end of the list.
Using the Graphical User Interface Keyboard Usage Keyboard Usage The GUI pops up different keyboards to modify data. Following are the usual keyboard keys: Left arrow moves the cursor one position to the left. Right arrow moves the cursor one position to the right. Up arrow increases the byte value by one. Down arrow decreases the byte value by one. Del deletes the value at the cursor position. Back deletes the value preceding the cursor position. OK completes data entry.
4 Selecting and Starting a Test A test can be created either by selecting the test from the Test Menu or by loading a previously saved configuration (refer to Load/Save on page 86 for more information). The EXpert VoIP Test Tools main menu offers the following test tools: Test Tools RTP on page 12 SIP on page 13 SCCP on page 15 H.323 on page 17 H.
Selecting and Starting a Test RTP RTP The RTP (Real Time Protocol) network test is a peer-to-peer test that measures the VoIP related parameters by streaming RTP packets between two endpoints. The RTP network test simulates VoIP traffic by streaming RTP packets between a Controller test set, which initiates the packet stream, and a Responder endpoint, which initiates its own stream to the Controller test set. The Controller sends an audio packet stream to the Responder.
Selecting and Starting a Test SIP SIP The SIP (Session Initiation Protocol) test measures the performance of Voice over IP (VoIP) services within a network. The test uses the SIP for call signaling and the RTP for digitally transporting the encoded audio. The SIP signaling portion of the test can be transported over UDP/TCP. SIP calls transported over UDP/TCP can be routed through a proxy server. The test's RTP functionality conforms to RFC 3550.
Selecting and Starting a Test SIP To select, configure, and start a SIP Test: 1. From the Main Menu, tap SIP. 2. From the Interface tab, configure the test interface parameters. See Interface on page 20. Ensure that the link is up in the status bar before proceeding to the next step. 3. From the Configuration tab, configure the SIP test. See Configuration (SIP) on page 26. 4. From the Controller Setup tab, configure the controller settings. See Controller Setup on page 42. 5.
Selecting and Starting a Test SCCP SCCP The SCCP (Skinny Client Control Protocol) service active test measures the performance of calls between two endpoints and a SCCP peer endpoint made through the Cisco Call Manager. In the SCCP test, the Controller acts as a calling endpoint (Caller), and the Responder acts as an answering endpoint (Callee). Both endpoints must be registered in the CallManager.
Selecting and Starting a Test SCCP 6. Press Start to start the test. 7. For results, refer to Summary (SCCP) on page 79, Call Quality Details on page 62, and Call Manager Statistics (SCCP) on page 60. 8. If required, tap the Report button to generate the report file of the results and statistics. Refer to Report on page 88 for more information.
Selecting and Starting a Test H.323 H.323 The H.323 test is a set-to-service test that measures call statistics for an H.323 call. It either makes a H.323 call to a given destination (directly to an endpoint, or through the gatekeepers and/or gateways that you specify) or answers a call made to it from an endpoint, and reports call setup and media path statistics. The test assumes that you know the destination (specified using an IP address, telephone number, or H.
Selecting and Starting a Test H.248/Megaco H.248/Megaco The H.248, the Media Gateway Control Protocol, is designed for distributed media gateway systems in which one or more centralized gateway controllers (Call Agents) control one or more media gateways. The VoIP H.248 test measures the performance of the services of H.248 based VoIP networks. It either initiates a call to or answers a call from other endpoints based on a telephone number. Performance is measured for both the call signaling path (H.
5 Test Setup The test setup offers the following tabs depending on the test: Tabs Test Page RTP SIP SCCP H.323 H.
Test Setup Interface Interface Note: The interface configuration parameters apply globally to all tests. From the Main Menu, tap on a test and the Interface tab. Test Interface Test Interface displays the network adapter(s) found on the EXFO’s platform. The first network adapter is selected by default.
Test Setup Interface Network Configuration Note: Network configuration is only possible when the Obtain IP from DHCP check box is cleared. Otherwise, when the Obtain IP from DHCP check box is selected, all parameters are not configurable and set to the values obtained through DHCP process. EXpert IP IP Address allows to enter the IP address for the Ethernet port: 0.0.0.1 to 223.255.255.255. Subnet Mask allows to enter the Subnet Mask for the Ethernet port: 0.0.0.1 to 255.255.255.255.
Test Setup Interface Ethernet Statistics The following Ethernet Statistics are displayed. Link Status displays the status of the link: Connected or Disconnected. Link Speed (Mbps) displays the speed of the Ethernet connection. Domain Name displays the name of the connected domain. Frame TX displays the total Ethernet frames transmitted. Frame RX displays the total Ethernet frames received. Duplex displays the Duplex mode of the link: half duplex, full duplex, or auto.
Test Setup Configuration (RTP) Configuration (RTP) From the Main Menu, tap RTP and the Configuration tab. Call Mode Select the call mode: Controller establishes a control communications channel with the Responder. It also sends the configuration details to the Responder. By default, the Controller is selected. Responder (default) is the endpoint, which initiates its own stream to the Controller test set. When Responder is selected, all other configuration parameters are disabled.
Test Setup Configuration (RTP) Codec represents the audio Codec (packet interval, payload size, and IP bandwidth) of the call simulated by the Controller. For Live Call media type: G.711 μlaw, and G.711 aLaw (default). For Synthesized Media type: G.711 μlaw, G.711 aLaw (default), G.722, G.723.1, G.726, G.728, G.729, G.729A, G.729B, and G.729AB. IP Media DSCP1 (IP Media Differentiated Services Code Point) describes a set of end-to-end QoS (Quality of Service) capabilities.
Test Setup Configuration (RTP) Record check box when selected (cleared by default) allows recording network VoIP traffic. The recorded file can be listened from the Load/Save option; refer to Save/Load - Record Audio Tab on page 87 for more information. Record is only available for Live Call media type. Capture Filters Available when the Utilities - Capture check box is selected.
Test Setup Configuration (SIP) Configuration (SIP) From the Main Menu, tap SIP and the Configuration tab. Peer Device EXFO Device: Select EXFO Device option on both the endpoints (i.e. Controller side and the Responder side) to run the SIP Network Active Test. Other Device (default): Select the Other Device option to run the service active test. In service active test, the test set can either act as controller or responder with the other end being the SIP endpoint.
Test Setup Configuration (SIP) Test Length Test Duration (sec) is the length of time for which the test will run in seconds: 1 to 600 seconds (default is 10 seconds). Test Length is not available when Stored Audio File media type is selected. Number of Packets is the number of RTP packets to exchange between the Controller and the Responder: 2 to 65535 (default is 500).
Test Setup Configuration (SIP) 28 Call Type selects the type of the call. For Peer Device set as EXFO Device, only SIP Call is available. In Network Active test, the supported call type is SIP Call. SIP Call attempts to establish a single call with a single audio session. SIP Options Request attempts a single “options” request method with the specified endpoint. Use this mode to test the availability of the SIP endpoint without making a complete call.
Test Setup Configuration (SIP) For Controller Call Mode: Disabled (default) doesn‘t use the PRACK method. Supported does not insist on usage of reliable provisional responses. However, the UAC supports the reliable provisional responses in case where the UAS needs to send one. The UAC creates a request by inserting a Supported header field into the request. Required insists on reliable delivery of provisional responses.
Test Setup Configuration (SIP) Destination IP address Only available when the Peer Device is set as EXFO Device. Manually Configure IP allows to configure the destination IP address manually. Auto Discovery List allows to select the IP address from a list of IP addresses discovered by the EXpert VoIP Test Tools. IP address allows to either enter the IP address manually (from 0.0.0.1 to 223.255.255.255) or from a list.
Test Setup Configuration (SCCP) Configuration (SCCP) From the Main Menu, tap SCCP and the Configuration tab. Call Mode Make Call: The Caller establishes a control communications channel with the Receiver. By default, Make Call is selected. Answer Call: It is the endpoint, which initiates its own stream to the Caller test set. Test Length Select the test length. Not available when Stored Audio File is selected as the media type.
Test Setup Configuration (SCCP) Codec selects the audio Codec (packet interval, payload size, and IP bandwidth) of the call simulated by the Caller. For Live Call media type: G.711 μlaw (default), and G.711 aLaw. For Audio File media type: G.711 μlaw (default), G.711 aLaw, G729, G.729A, G.729B, and G.729AB. For Synthesized Media type: G.711 μlaw (default), G.711 aLaw, G.722, G.723.1, G.726, G.728, G.729, G.729A, G.729B, and G.729AB.
Test Setup Configuration (SCCP) Utilities Traceroute check box when selected (default) calculates the performance of the network path between two endpoints of a running test every time the test is run. Capture check box when selected (cleared by default) captures the RTP packet stream for the running test. Once test is completed, the capture file is available and can be opened in the Wireshark application from the Load/Save option; refer to Packet Capture on page 87 for more information.
Test Setup Configuration (H.323) Configuration (H.323) From the Main Menu, tap H.323 and the Configuration tab. Call Mode Make Call: The Caller establishes a control communications channel with the Receiver. By default, Make Call is selected. Answer Call: It is the endpoint, which initiates its own stream to the Caller test set. Call Information Media Type Live Call (default) specifies that actual call is established between the Caller and the Callee.
Test Setup Configuration (H.323) DSCP1 IP Media DSCP (IP Media Differentiated Services Code Point) describes a set of end-to-end QoS (Quality of Service) capabilities. End-to-end QoS is the ability of the network to deliver service required by the specific network traffic from one end of the network to the other end. The accepted range is from 0 (default) to 63. Call Duration Call Duration (sec) is the length of time for the H.323 session. The accepted range is from 1 to 3600 seconds (default is 10).
Test Setup Configuration (H.323) Transport Address specifies the call destination’s transport address. Enter the IP address with Q.931 TCP server port number or DNS name with Q.931 TCP server port number of the called H.323 terminal. The TCP server port number is optional. The format used for destination address transport is [:port]. The default port number is 1720. For example 192.168.1.150 is equivalent to 192.168.1.150:1720, and yourname.com is equivalent to yourname.com:1720.
Test Setup Configuration (H.323) Capture Filters Available when the Utilities - Capture check box is selected. EXpert IP IP Address check box when selected (cleared by default) represents the Peer IP Address that acts as a filter that will capture packets having the source or destination IP address matching the specified Peer IP address. The accepted range is from 0.0.0.1 to 223.225.225.225.
Test Setup Configuration (H.248/Megaco) Configuration (H.248/Megaco) From the Main Menu, tap H.248/Megaco and the Configuration tab. Call Mode Make Call (default): The Caller establishes a control communications channel with the Receiver. By default, Make Call is selected. Answer Call: It is the endpoint, which initiates its own stream to the Caller test set. Test Length Select the test length.
Test Setup Configuration (H.248/Megaco) H.248 DSCP1 (H.248 Differentiated Services Code Point) describe a set of QoS (Quality of service) capabilities. The accepted range is from 0 (default) to 63. IP Media DSCP1 (IP Media Differentiated Services Code Point) describes a set of end-to-end QoS (Quality of Service) capabilities. End-to-end QoS is the ability of the network to deliver service required by the specific network traffic from one end of the network to the other end.
Test Setup Configuration (H.248/Megaco) Capture Filters Available when the Utilities - Capture check box is selected. 40 IP Address check box when selected (cleared by default) represents the Peer IP Address that acts as a filter that will capture packets having the source or destination IP address matching the specified Peer IP address. The accepted range is from 0.0.0.1 to 223.225.225.225.
Test Setup Controller Setup Controller Setup Note: Only available when Controller is selected as the call mode (see Call Mode on page 26). From the Main Menu, tap SIP and the Controller Setup tab. Controller Parameters EXpert IP User Name is the name used for the SIP URL associated with the Controller. A maximum 50 characters are allowed. Signaling Port is the signaling port of the Controller: 1000 to 65535 (default is 5060).
Test Setup Controller Setup Proxy Settings Force Proxy If the Force Proxy check box is selected, the test routes all requests associated with each call through the specified proxy server. The test ignores any Record-Route or contact information contained in the final response to the initial “SIP invite” request method. If the Force Proxy check box is cleared (default), the test routes subsequent requests in one of the following ways.
Test Setup Controller Setup Registration Parameters Note: Available when Other Device is selected as Peer Device (see Peer Device on page 26). EXpert IP Specify Registrar specifies SIP Registrar or Registration-Server. When the Specify Registrar check box is selected (cleared by default), SIP registration requests are accepted and processed by the specified SIP Registrar.
Test Setup Responder Setup Responder Setup Note: Not available when both EXFO Device (see Peer Device on page 26) and Responder (see Call Mode on page 26) are selected. From the Main Menu, tap SIP and the Responder Setup tab. Responder Parameters 44 User Name is the name of the Responder, which is used for the SIP URL associated with the Responder. A maximum of 50 characters are allowed. Domain Name is the DNS name used as the host portion of the Responder’s SIP URL.
Test Setup Responder Setup Proxy Settings Force Proxy If the Force Proxy check box is selected, the test routes all requests associated with each call through the specified proxy server. The Responder ignores any Record-Route or contact information contained in the final response to the initial “SIP invite” request method. If the Force Proxy check box is cleared (default), the test routes subsequent requests in one of the following ways.
Test Setup Responder Setup Registration Parameters Note: Available when Other Device is selected as Peer Device (see Peer Device on page 26). 46 Register check box when selected (cleared by default) specifies that the SIP-Responder will register with a SIP Proxy (or SIP Registrar -if specified) during every test run. Specify Registrar specifies SIP Registrar or Registration-Server.
Test Setup Parameters (SCCP) Parameters (SCCP) From the Main Menu, tap SCCP and the Parameters tab. Call Parameters Call Manager IP Address/DNS allows to enter the IP Address or DNS of the Call Manager. IP address can be in a dotted decimal notation (for example, 192.168.34.5) or in a DNS name (for example, cm.acme.com) format. A maximum of 56 characters are allowed. Local MAC Address allows to enter the MAC Address associated with the Callee as specified in the CallManager registration tables.
Test Setup Parameters (H.323) Parameters (H.323) From the Main Menu, tap H.323 and the Parameters tab. Local Parameters Gatekeeper IP Address/DNS allows to enter the IP address, or DNS name, or optionally, the UDP RAS port number of the H.323 Gatekeeper. A maximum of 56 characters are allowed. Gatekeeper Port allows to enter the port number of the Gatekeeper: 1000 to 65535 (default is 1719).
Test Setup Parameters (H.323) Media Parameters EXpert IP Media Mode defines the flow direction of the RTP packets (media). Send and Receive (default) specifies that the RTP packets flows in both directions. For example, if Controller is set in Send and receive media mode then it will send RTP Packets to Responder and also it will receive the RTP packets from Responder. Receive Only specifies that the RTP packets flows only in one direction, i.e.
Test Setup Parameters (H.248/Megaco) Parameters (H.248/Megaco) From the Main Menu, tap H.248/Megaco and the Parameters tab. Local Parameters 50 H.248 End Point Name allows to enter the H.248 End Point Name. The End Point Name should be registered with the end point location (i.e. IP address) and the phone number present in the Call Agent or Media Gateway Controller. For example, if the End Point Name is registered with the abc/1 and abd/1, then you can enter abc and abd in the H.
Test Setup Threshold Threshold The threshold is used to specify the Pass and Fail values for the test. From the Main Menu, tap any test and the Threshold tab. VoIP Pass/Fail Thresholds EXpert IP Select All enables all the threshold parameters. Maximum Jitter (ms): Enter the maximum Jitter value: 0 (default) to 125 milliseconds. The test fails if the average jitter is greater than the configured value. Maximum Latency (ms): Enter the Maximum Latency value: 0 (default) to 600 milliseconds.
6 Test Results The rest Results menu offers the following Result Summary tabs depending on the test: Tabs Test RTP Call Details SIP SCCP H.323 H.
Test Results Call Details Call Details From the Tests menu, tap Results and the Call Details tab. Test Results Endpoint IP Address displays the IP Address of the endpoint device. Peer URL displays the URL of the endpoint device. Response is the method and response string of the last SIP message received by the test. Response Code is the SIP Response code that the destination returned during the last SIP transaction.
Test Results Call Details Registration Time (ms) indicates the time in milliseconds to complete a SIP “register request” method. Authentication Time (ms) indicates the time in milliseconds to complete the proxy authentication during call initiation. Note: The Authentication Time result is valid only, if SIP Proxy Registration Only or SIP Call After Proxy Registration is selected under Call Type in Call Information in the Configuration tab.
Test Results Call Manager Statistics (H.248/Megaco) Call Manager Statistics (H.248/Megaco) Displays the performance statistics of the Call Manager. From the Tests menu, tap Results and the Call Manager Statistics tab. Result Summary Metrics Peer Phone Number is the phone number used by the endpoint in Caller mode. Peer IP Address is the IP address of the peer endpoint. Note: RTCP (Real-time Transport Control Protocol) provides out of band statistics and control information for an RTP flow.
Test Results Call Manager Statistics (H.248/Megaco) EXpert IP Post Pickup Delay (ms) is the post pickup delay measured by the Caller in milliseconds. This value is the difference between the transmission of the Off Hook Notification message by the test and the reception of the first media (RTP) packet from the endpoint. A negative value indicates that the media arrived early (audio clipping). Total Dial Delay (ms) is the time in milliseconds to complete the call initiation as seen by the Caller.
Test Results Call Manager Statistics (H.323) Call Manager Statistics (H.323) Displays the performance statistics of the Call Manager. From the Tests menu, tap Results and the Call Manager Statistics tab. General Results Endpoint Display Information is provided by the peer endpoint for the call. Endpoint Phone Number is the called number used for this call. Receiving Audio Codec is the Audio Codec used for the inbound audio channel from the H.323 endpoint to the Caller.
Test Results Call Manager Statistics (H.323) Ring Duration (ms) is the length of time in milliseconds for a called terminal to answer the call after it rings. Media Duration (sec) is the length of time in seconds for the duration of the test. Gatekeeper Results EXpert IP Gatekeeper Address displays the Gatekeeper DNS or IP Address through which the call happens. Discovery Delay (ms) is the length of time in milliseconds for the Gatekeeper Discovery Delay.
Test Results Call Manager Statistics (SCCP) Call Manager Statistics (SCCP) Displays the performance statistics of the Call Manager. From the Tests menu, tap Results and the Call Manager Statistics tab. Call Statistics Assigned Phone Number is the phone number assigned to the endpoint (caller) by the Call Manager during registration. Peer Phone Number is the phone number used by the endpoint in Caller mode to reach the peer endpoint.
Test Results Call Manager Statistics (SCCP) EXpert IP Post Dial Delay is the time in milliseconds between the transmission of the last dialed digit to the CallManager and the reception of the Alerting tone message. Post Pickup Delay is the time in milliseconds between the reception of the SCCP Open Receive Channel “Ack” message and the reception of the first media RTP packet.
Test Results Call Quality Details Call Quality Details From the Tests menu, tap Results and the Call Quality Details tab. MOS / R-Factor EXFO Conversational MOS is the Voice Quality Conversational Mean Opinion Score (MOS) score for the incoming RTP stream. User R-Factor is the Voice Quality R-factor for the incoming RTP stream. RTP Statistics 62 Packet Count is the number of good packets received from endpoint. Lost Packets is the Number of lost packets in the incoming RTP stream.
Test Results Call Quality Details EXpert IP Out-of-Order Packets is the number of RTP packets that were received out of order. This value is calculated using the jitter buffer length input parameter value. Duplicate Packets is the number of duplicate packets received and dropped by the local endpoint. No. of Loss Periods is the count for the number of periods of lost RTP packets.
Test Results Call Quality Details Degradation Factors Codec (%) is the percentage of MOS score degradation that was due to the codec type in the incoming RTP stream. Latency (%) is the percentage of MOS score degradation that was due to the delay in the incoming and outgoing RTP stream. Packet Loss (%) is the percentage of MOS score degradation that was due to the packet loss in the incoming RTP stream.
Test Results Controller Statistics Controller Statistics Displays the result information for packets that travelled from the responder to the controller. Note: The Controller Statistics tab is only available when Responder is selected under the Call Mode from the Configuration tab. From the Tests menu, tap Results and the Controller Statistics tab.
Test Results Controller Statistics 66 Total Lost Packets (%) is the percentage of RTP packets that were either lost or late. Out-of-Order Packets is the number of RTP packets that were received out of order. This value is calculated using the jitter buffer length input parameter value. Duplicate Packets is the number of duplicate test packets received and dropped by the Controller. No. of Loss Periods is the count for the number of periods of lost RTP packets.
Test Results Controller Statistics Degradation Factors Codec (%) is the percentage of MOS score degradation that was due to the codec type in the RTP stream from the Responder to the Controller. Latency (%) is the percentage of MOS score degradation that was due to the delay in the RTP stream from the Responder to the Controller. Packet Loss (%) is the percentage of MOS score degradation that was due to the packet loss in the RTP stream from the Responder to the Controller.
Test Results Responder Statistics Responder Statistics Displays the result information for packets that travelled from the controller to the responder. From the Tests menu, tap Results and the Responder Statistics tab. MOS / R-factor EXFO Conversion MOS is the Voice Quality Conversational Mean Opinion Score (MOS) score for the RTP stream from the Controller to Responder. User R-Factor is the Voice Quality R-factor for the RTP stream from the Controller to Responder.
Test Results Responder Statistics EXpert IP Out-of-Order Packets is the number of RTP packets that were received out of order. This value is calculated using the jitter buffer length input parameter value. Duplicate Packets is the number of duplicate test packets received and dropped by the Responder. No. of Loss Periods is the count for the number of periods of lost RTP packets.
Test Results Responder Statistics Degradation Factors Codec (%) is the percentage of MOS score degradation that was due to the codec type in the RTP stream from the Controller to Responder. Latency (%) is the percentage of MOS score degradation that was due to the delay in the RTP stream from the Controller to Responder. Packet Loss (%) is the percentage of MOS score degradation that was due to the packet loss in the RTP stream from the Controller to Responder.
Test Results Summary (H.248/Megaco) Summary (H.248/Megaco) From the Tests menu, tap Results and the Summary tab. Status displays the Status of the test. Once the test is completed, the Pass/Fail verdict is displayed. Running: test is running. Stopped: test is interrupted; test is stopped before the set time. Completed: test is completed successfully. Message displays one of the following error messages when the test fails. Call Agent DNS Lookup Failure. Call Agent host unreachable.
Test Results Summary (H.248/Megaco) Voice Quality Summary Metrics MOS is the Mean Opinion Score for the RTP stream (packets). If Make Call is selected as Call Mode, in the Configuration tab, then the value is calculated for RTP Stream (packets) from Callee to Caller. If Answer Call is selected as Call Mode in the Configuration tab, then the value is calculated for RTP Stream (packets) from Caller to Callee. R-Factor: The Voice Quality R-factor for the RTP stream.
Test Results Summary (H.323) Summary (H.323) From the Tests menu, tap Results and the Summary tab. Status displays the Status of the test. Once the test is completed, the Pass/Fail verdict is displayed. Running: test is running. Stopped: test is interrupted; test is stopped before the set time. Completed: test is completed successfully. Message displays one of the following error messages when the test fails. DNS gatekeeper lookup failed. Gatekeeper discovery failed.
Test Results Summary (H.323) Voice Quality Summary Metrics MOS is the Mean Opinion Score for the RTP stream (packets). If Make Call is selected as Call Mode in the Configuration tab, then the value is calculated for RTP Stream (packets) from Callee to Caller. If Answer Call is selected as Call Mode in the Configuration tab, then the value is calculated for RTP Stream (packets) from Caller to Callee. R-Factor: The Voice Quality R-factor for the RTP stream.
Test Results Summary (H.323) Packet Loss (%) is the percentage of lost, late, and early packets traveling on the network. If Make Call is selected as the Call Mode in the Configuration tab, then the value calculated is from Callee to Caller. If Answer Call is selected as the Call Mode in the Configuration tab, then the value calculated is from Caller to Callee. Latency is the length of time in milliseconds for the minimum and maximum latency.
Test Results Summary (RTP) Summary (RTP) From the Tests menu, tap Results and the Summary tab. Status displays the Status of the test. Once the test is completed, the Pass/Fail verdict is displayed. Running: test is running. Stopped: test is interrupted; test is stopped before the set time. Completed: test is completed successfully. Message displays one of the following error messages when the test fails. DNS lookup failed. No Response from Responder. Responder unreachable.
Test Results Summary (RTP) Voice Quality Summary Metrics MOS: The Mean Opinion Score (MOS) is the value calculated for RTP Stream (packets). If Controller is selected as Call Mode in the Configuration tab, then the value is calculated for RTP Stream (packets) from Responder to Controller. If Responder is selected as Call Mode in the Configuration tab, then the value is calculated for RTP Stream (packets) from Controller to Responder. R-Factor: The Voice Quality R-factor for the RTP stream.
Test Results Summary (RTP) Packet Loss (%) Threshold: The threshold value set from the threshold configuration page. No Packets Sent: The number of RTP packets that were transmitted to other peer. Latency Length of time in milliseconds for the minimum and maximum latency. Traceroute 78 Hops to Controller is the number of hops from the Responder to the Controller. Hops to Responder is the number of hops from the Controller to the Responder.
Test Results Summary (SCCP) Summary (SCCP) From the Tests menu, tap Results and the Summary tab. Status displays the Status of the test. Once the test is completed, the Pass/Fail verdict is displayed. Running: test is running. Stopped: test is interrupted; test is stopped before the set time. Completed: test is completed successfully. Message displays one of the following error messages when the test fails. DNS lookup failed. Host Unreachable. No Response. Request Refused.
Test Results Summary (SCCP) R-Factor is the Voice Quality R-factor for the RTP stream. If Make Call is selected as Call Mode in the Configuration tab, then the value is calculated for RTP Stream (packets) from Callee to Caller. If Answer Call is selected as Call Mode in the Configuration tab, then the value is calculated for RTP Stream (packets) from Caller to Callee. Jitter (ms) is the average length of time in milliseconds of the jitter of the RTP packets of incoming and outgoing RTP stream.
Test Results Summary (SIP) Summary (SIP) From the Tests menu, tap Results and the Summary tab. Status displays the Status of the test. Once the test is completed, the Pass/Fail verdict is displayed. Running: test is running. Stopped: test is interrupted; test is stopped before the set time. Completed: test is completed successfully. Message displays one of the following error messages when the test fails. Call initiate timeout. Call initiate failure. Call terminate timeout.
Test Results Summary (SIP) Voice Quality Summary Metrics MOS is the Mean Opinion Score (MOS) for the audio RTP stream (packets). If Controller is selected as Call Mode in the Configuration tab, then the value is calculated for RTP Stream (packets) from Responder to Controller. If Responder is selected as Call Mode in the Configuration tab, then the value is calculated for RTP Stream (packets) from Controller to Responder. R-Factor is the Voice Quality R-factor for the RTP stream.
Test Results Summary (SIP) EXpert IP Packet Loss (%) - Threshold: The threshold value set from the threshold configuration page. Packet Sent is the number of Packets transmitted to other peer. Latency is the length of time in milliseconds for the minimum and maximum latency.
7 Test Control This chapter describes the test control buttons available on the top-right navigation bar of the application. Start/Stop Button The Start/Stop button allows to manually start or stop the test. EXpert IP Start: Tap the Start button to start the test. Start is available when the test is not running. Stop: Tap the Stop button to stop the test. Stop is available when the test is running.
Test Control Load/Save Load/Save Configurations Tab The Configuration tab allows to save, load, and delete a test configuration Tap the Load/Save button, and the Configuration tab. Restore Factory Default restores the configuration parameters of all the tests to factory default setting. To save a configuration: 1. Select the media where the file will be saved: Internal Flash (default) or a removable drive (USB media for example) if present. 2. Type the name of the file to be saved (File Name) if needed. 3.
Test Control Load/Save Record Audio Tab The Record Audio tab allows to listen or delete an audio file. Tap the Save/Load button, and the Record Audio tab. Delete Audio File button deletes the selected audio file from the list. Listen Audio File button plays the selected audio file from the list. Packet Capture The Packet Capture tab allows to open or delete a captured packet file. Tap the Save/Load button, and the Packet Capture tab.
Test Control Report Report Config/Save Tab The Config/Save tab allows to configure the report parameters and generate/save the report. Tap the Report button, and the Config/Save tab. The report contains all information about the current test including its setup and results. Report Content parameters are used to identify the report and are not mandatory. Up to 50 characters are allowed for each parameter. Report Header could be the company name.
Test Control Report Turn on Report Generation Prompt check box when selected (default) displays a pop-up every time a test case is stopped or completed to ask if a report generation is desired. Format is the file format for the report: PDF (default), Text, HTML, and CSV. Logo check box when selected (default) allows to include a logo to the report. Only available with the PDF, and HTML file format. Select the logo picture that will be displayed on the report.
8 Troubleshooting Solving Common Problems Before calling EXFO’s technical support, please read the following common problems that can occur and their respective solution. Problem Possible Cause Solution SIP call not working. Signaling port may not be correct. Check whether signaling port is configured correctly. SCCP call not working. SCCP Call Manager may not Check whether Call be available. Manager is up. H.323 call not working. H.323 ID of destination phone may not be correct. Check whether H.
Troubleshooting Contacting the Technical Support Group Contacting the Technical Support Group To obtain after-sales service or technical support for this product, contact EXFO at one of the following numbers. The Technical Support Group is available to take your calls from Monday to Friday, 8:00 a.m. to 7:00 p.m. (Eastern Time in North America). Technical Support Group 400 Godin Avenue Quebec (Quebec) G1M 2K2 CANADA 1 866 683-0155 (USA and Canada) Tel.: 1 418 683-5498 Fax: 1 418 683-9224 support@exfo.
A Glossary Acronym List ? Help AC Alternating Current ARP Address Resolution Protocol ASCII American Standard Code for Information Interchange bps Bit Per Second CE European Conformity DHCP Dynamic Host Configuration Protocol DNS Domain Name Server FCC Federal Communications Commission A B C D F EXpert IP 93
Glossary Acronym List G GUI Graphical User Interface ID Identification IEEE Institute of Electrical & Electronics Engineers IP Internet Protocol ISO International Organization for Standardization LAN Local Area Network LED Light-Emitting Diode m Minute m Meter MAC Media Access Control Mbps Megabit Per Second MOS Mean Opinion Score ms millisecond PRACK Provisional Response ACKnowledgement I L M P 94 EXpert VoIP Test Tools
Glossary Acronym List Q QoS Quality of Service RFC Request For Comments RJ-45 Registered Jack 45 RMA Return Merchandise Authorization RTP Real Time Protocol RX Receive SCCP Skinny Client Control Protocol SIP Session Initiation Protocol TCP Transport Control Protocol TTL Time To Live TX Transmit UAC User Agent Client UAS User Agent Server UDP User Data Protocol URL Uniform Resource Locator R S T U EXpert IP 95
Glossary Acronym List μs microsecond USB Universal Serial bus VLAN Virtual Local Area Network VoIP Voice over Internet Protocol V 96 EXpert VoIP Test Tools
Index Index A About button ................................................ 7 Admission Delay.......................................... 59 after-sales service ........................................ 92 Answer Call ........................................... 32, 39 Apply........................................................... 21 Arrow buttons............................................... 8 Assigned Phone Number ............................. 60 Authentication Password ......................
Index Ethernet Statistics ....................................... 22 EXFO Conversational MOS..................... 62, 65 EXFO Conversion MOS ................................ 68 Exit button .................................................... 7 F Frame RX..................................................... 22 Frame TX ..................................................... 22 G Gatekeeper Address .................................... 59 Gatekeeper IP Address/DNS......................... 49 Gatekeeper Port .
Index N Network Bandwidth .................................... 74 No of Packets Sent ...................................... 54 No Packets Sent........................................... 78 No. of Loss Periods .......................... 63, 66, 69 Number of Notified Entities......................... 57 Number of Packets .......................... 27, 32, 39 O Obtain IP from DHCP................................... 20 Off Hook Time....................................... 56, 60 Option Response Time...........
Index Sending Audio Codec.................................. 58 Signaling Port........................................ 42, 45 SIP Call ........................................................ 28 SIP Call After Proxy Registration .................. 28 SIP DSCP...................................................... 29 SIP Options Request .................................... 28 SIP Proxy Registration.................................. 28 Software options...........................................
P/N: 1067245 www.EXFO.com · info@exfo.com CORPORATE HEADQUARTERS 400 Godin Avenue Quebec (Quebec) G1M 2K2 CANADA Tel.: 1 418 683-0211 · Fax: 1 418 683-2170 EXFO AMERICA 3400 Waterview Parkway Suite 100 Richardson, TX 75080 USA Tel.: 1 972-761-9271 · Fax: 1 972-761-9067 EXFO EUROPE Winchester House, School Lane Chandlers Ford, Hampshire S053 4DG ENGLAND Tel.: +44 2380 246 800 · Fax: +44 2380 246 801 EXFO ASIA-PACIFIC 62 Ubi Road 1, #09-01/02 Oxley Bizhub 2 SINGAPORE 408734 Tel.