Remote Monitoring

Remote Monitoring Overview

Each Cisco Unified IP Conference Phone has a web page from which you can view a variety of information about the conference phone, including:

  • Device information

  • Network configuration information

  • Ethernet information

  • Device logs

  • Streaming statistics

This chapter describes the information that you can obtain from the conference phone web page. You can use this information to remotely monitor the operation of a conference phone and to assist with troubleshooting.

You can also obtain much of this information directly from a conference phone. For more information, see Model Information, Status, and Statistics.

For more information about troubleshooting the conference phone, see Troubleshooting and Maintenance.

Access Web Page


Note

If you cannot access the web page, it may be disabled. See the Control Web Page Access section for more information.


Procedure


Step 1

Obtain the IP address of the conference phone using one of these methods:

  • From Cisco Unified Communications Manager Administration, choose Device > Phone. Enter search criteria to locate the conference phone, and then click the conference phone name. Conference phones registered with Cisco Unified Communications Manager display the IP address at the top of the Phone Configuration web page.

  • On the conference phone, choose Apps > Settings > Network Configuration. Then, scroll to the IP Address option.

Step 2

Open a web browser and enter the following URL, where IP address is the IP address of the conference phone: http://<IP_address>


Web Page Information

The web page for a conference phone includes these hyperlinks:

  • Device Information: Displays device settings and related information for the conference phone.

  • Network Configuration: Displays network configuration information and information about other conference phone settings.

  • Ethernet Information: Displays network statistics.

  • Device Logging: Displays messages that might be useful to Cisco TAC if you require assistance with troubleshooting.

  • Streaming Statistics: Displays call statistics.

Control Web Page Access

For security purposes, you may choose to prevent access to the web pages for a conference phone. If you do so, you will prevent access to the web pages that are described in this chapter and to the Cisco Unified Communications Manager Self Care Portal.

To enable or disable access to the web pages for a conference phone, follow these steps from Cisco Unified Communications Manager Administration:

Procedure


Step 1

Choose Device > Phone.

Step 2

Specify the criteria to find the phone and click Find, or click Find to display a list of all phones.

Step 3

Click the device name to open the Phone Configuration web page for the device.

Step 4

In the Product Specific Configuration Layout area, from the Web Access drop-down list, choose Enabled or Disabled.

Step 5

Click Update.

Note 

Some features, such as Cisco Quality Report Tool, do not function properly without access to the conference phone web pages. Disabling web access also affects any serviceability application that relies on web access, such as CiscoWorks.


Device Information Area

The Device Information area on a conference phone web page displays device settings and related information for the conference phone. The following table describes these items.

To display the Device Information area, access the web page for the conference phone as described in the Access Web Page section, and then click the Device Information hyperlink.

Table 1. Device Information Area Items

Item

MAC Address

Media Access Control (MAC) address of the conference phone.

Host Name

Unique, fixed name that is automatically assigned to the conference phone based on its MAC address.

DN

Directory number assigned to the conference phone.

Version

Version of the firmware running on the conference phone.

Hardware Revision

Revision value of the conference phone hardware.

Serial Number

Serial number of the conference phone.

Model Number

Model number of the conference phone.

Message Waiting

Indicates if there is a voice message waiting for this conference phone.

UDI

Displays the following Cisco Unique Device Identifier (UDI) information about the conference phone:
  • Device Type: Indicates hardware type. For example, phone displays for all phone models

  • Device Description: Displays the name of the conference phone associated with the indicated model type

  • Product Identifier: Specifies the conference phone model

  • Serial Number: Displays the conference phone unique serial number

Time

Time obtained from the Date/Time Group in Cisco Unified Communications Manager to which the conference phone belongs.

Time Zone

Time zone obtained from the Date/Time Group in Cisco Unified Communications Manager to which the conference phone belongs.

Date

Date obtained from the Date/Time Group in Cisco Unified Communications Manager to which the conference phone belongs.

Network Configuration Area

The Network Configuration area on a conference phone web page displays network configuration information and information about other conference phone settings. The following table describes this information.

You can view and set many of these items from the Network Configuration Menu and the Device Configuration Menu on the conference phone.

To display the Network Configuration area, access the web page for the conference phone as described in the Access Web Page section, and then click the Network Configuration hyperlink.

Item

Description

DHCP Enabled

Indicates whether DHCP is being used by the conference phone.

MAC Address

MAC address of the conference phone.

Host Name

Host name that the DHCP server assigned to the conference phone.

IP Address

IP address of the conference phone.

Subnet Mask

IP address of the subnet mask used by the conference phone.

Default Router 1

Default router used by the conference phone (Default Router 1).

Domain Name

Name of the Domain Name System (DNS) domain in which the conference phone resides.

DNS Server 1–5

Primary Domain Name System (DNS) server (DNS Server 1) and optional backup DNS servers (DNS Server 2–5) used by the conference phone.

Operational VLAN ID

Auxiliary Virtual Local Area Network (VLAN) configured on a Cisco Catalyst switch in which the conference phone is a member.

Admin. VLAN ID

Auxiliary VLAN in which the conference phone is a member.

TFTP Server 1

Primary Trivial File Transfer Protocol (TFTP) server used by the conference phone.

TFTP Server 2

Optional backup TFTP server that the conference phone uses if the primary TFTP server is unavailable.

Alternate TFTP

Indicates whether the conference phone is using an alternate TFTP server.

Ethernet Configuration

Speed and duplex of the Ethernet port (labeled LAN on the conference phone).

CallManager 1–5

Host names or IP addresses, in prioritized order, of the Cisco Unified Communications Manager servers with which the conference phone can register. An item can also show the IP address of a Survivable Remote Site Telephony (SRST) router that is capable of providing limited Cisco Unified Communications Manager functionality, if such a router is available.

For an available server, an item will show the Cisco Unified Communications Manager server IP address and one of the following states:
  • Active: Cisco Unified Communications Manager server from which the conference phone is currently receiving call-processing services.

  • Standby: Cisco Unified Communications Manager server to which the conference phone switches if the current server becomes unavailable.

  • Blank: No current connection to this Cisco Unified Communications Manager.

An option may also include the SRST designation, which indicates an SRST router capable of providing Cisco Unified Communications Manager functionality with a limited feature set. This router assumes control of call processing if all other Cisco Unified Communications Manager servers become unreachable. The SRST Cisco Unified Communications Manager always appears last in the list of servers, even if it is active. You configure the SRST router address in the Device Pool section in Cisco Unified Communications Manager.

Information URL

URL of the help text that appears on the conference phone.

Services URL

URL of the server from which the conference phone obtains conference phone services.

Directories URL

URL of the server from which the conference phone obtains directory information.

Messages URL

URL of the server from which the conference phone obtains message services.

Authentication URL

URL that the conference phone uses to validate requests made to the conference phone web server.

Proxy Server URL

URL of proxy server, which makes HTTP requests to non-local host addresses on behalf of the conference phone HTTP client and provides responses from the non-local host to the conference phone HTTP client.

Idle URL

URL that the conference phone displays when the conference phone has not been used for the time specified by Idle URL Time and no menu is open.

Idle URL Time

Number of seconds that the conference phone has not been used and no menu is open before the XML service specified by Idle URL is activated.

User Locale

User locale associated with the conference phone user. Identifies a set of detailed information to support users, including language, font, date and time formatting, and alphanumeric keyboard text information.

User Locale Version

Version of the user locale loaded on the conference phone.

User Locale Char Set

Version of the character set that the conference phone uses for the user locale.

Network Locale

Network locale associated with the conference phone user. Identifies a set of detailed information to support the conference phone in a specific location, including definitions of the tones and cadences used by the conference phone.

Network Locale Version

Version of the network locale loaded on the conference phone.

DSCP For Call Control

DSCP IP classification for call control signaling.

DSCP For Configuration

DSCP IP classification for any conference phone configuration transfer.

DSCP For Services

DSCP IP classification for conference phone-based services.

Web Access Enabled

Indicates whether web access is enabled (Yes) or disabled (No) for the conference phone.

Ethernet Information Area

The Ethernet Information area on a conference phone web page provides information about network traffic on the conference phone, such as:
  • Ethernet traffic

  • Network traffic to and from the PC port on the conference phone

  • Network traffic to and from the network port on the conference phone

To display the Ethernet Information area, access the web page for the conference phone as described in the Access Web Page section, and then click the Ethernet Information hyperlink.

The following table describes the information in the Ethernet Information area.

Table 2. Ethernet Information Area Items

Item

Description

Rx error

Total number of FCS error packets or Align error packets received.

Rx PacketNoDes

Total number of shed packets caused by no DMA descriptor.

Rx Overruns

Total number of received packets dropped because of buffer overruns.

Rx alignErr

Total number of packets received between 64 and 1522 bytes in length that have bad FCS errors.

Rx length error

Number of packets discarded due to improper length.

Rx symbol error

Number of valid length packets received that have at least one invalid data symbol.

Rx CRC Errors

Total number of packets received with CRC failed.

Rx Broadcasts

Number of broadcast packets received by the conference phone.

Rx Multicasts

Total number of multicast packets received by the conference phone.

Rx fail filter

Total number of packets received by the conference phone that failed.

Rx VLAN

Total number of packets received on the Virtual Local Area Network.

Rx control frames

Total number of control frames received.

Rx unicast

Total number of unicast packets received by the conference phone.

Tx error

Total number of FCS error packets or Align error packets transmitted by the conference phone.

Tx no descriptor

Total number of transmit packets dropped because no descriptor was specified.

Tx fifoUnderrun

Total number of transmit packets dropped because of fifo underrun.

Tx lateCollision

Number of times that collisions occurred later than 512 bit times after the start of packet transmission.

Tx Excessive Collisions

Total number of packets that could not be sent because of network congestion.

Tx excessDefer

Total number of packets delayed from transmitting due to medium being busy.

Tx Deferred Abort

Total number of transmit packets aborted.

Tx Collisions

Total number of collisions that occurred while a packet was being transmitted.

Event send failed

Total number of packets that failed to transmit.

Event Rx packet send failed

Total number of packets that were not received.

Tx excessLength

Total number of packets not transmitted because the packet experienced 16 transmission attempts.

Rx totalPkt

Total number of packets received by the conference phone.

Packet Transmitted

Total number of packets transmitted by the conference phone.

Rcvr Octets

Total number of octets received by the conference phone.

Sender Octets

Total number of octets sent by the conference phone.

Device Logs Area

The Device Logs area on a conference phone web page provides information you can use to help monitor and troubleshoot the conference phone. It includes debug and error messages received on the conference phone that might be useful to Cisco TAC if you require assistance with troubleshooting.

To display device logs, access the web page for the conference phone as described in the Access Web Page section, and then click the Device Logging hyperlink. In the File Download dialog box, click Open to view the device logs, or click Save to save the logs to a specific location.

Streaming Statistics Area

A conference phone can stream information to and from up to three devices simultaneously. A conference phone streams information when it is on a call or running a service that sends or receives audio or data.

The Streaming Statistics area on a conference phone web page provides information about the streams. Most calls use only one stream (Stream 1), but some calls use two or three streams. For example, a barged call uses Stream 1 and Stream 2.

To display the Streaming Statistics area, access the web page for the conference phone as described in the Access Web Page section, and then click the Streaming Statistics hyperlink.

The following table describes the items in the Streaming Statistics areas.

Table 3. Streaming Statistics Area Items

Item

Description

Remote Address

IP address and UDP port of the stream.

Local Address

IP address and UDP port of the conference phone.

Start Time

Internal time stamp indicating when Cisco Unified Communications Manager requested that the conference phone start transmitting packets.

Codec Type

Type of voice stream received or transmitted (RTP streaming audio): G.729, G.711 u-law, G.711 A-law, G.722, or Lin16k.

Payload Size

Size of voice packets, in milliseconds, in the receiving or transmitting voice stream (RTP streaming audio).

Rcvr Packets

Number of RTP voice packets received since voice stream was opened.

Note 

This number is not necessarily identical to the number of RTP voice packets received since the call began because the call might have been placed on hold.

Rcvr Lost Packets

Missing RTP packets (lost in transit).

Rcvr Octets

Number of bytes of voice packets received since voice stream was opened.

Rx Expected Pkts

The expected number of packets received for the local conference phone.

Last Rx Seq No

The sequence number of the last RTP packet received.

Most recent Rx SSRC

The Synchronization Source field of the last RTP packet received.

Avg Jitter

Estimated average RTP packet jitter (dynamic delay that a packet encounters when going through the network) observed since the receiving voice stream was opened.

Max Jitter

Maximum jitter observed since the receiving voice stream was opened.

Sender Packets

Number of RTP voice packets transmitted since voice stream was opened.

Note 

This number is not necessarily identical to the number of RTP voice packets transmitted since the call began because the call might have been placed on hold.

Sender Octets

Number of bytes of voice packets transmitted since voice stream was opened.