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Cisco Packet Telephony Center Monitoring and Troubleshooting

Cisco Packet Telephony Center Monitoring & Troubleshooting

Data Sheet


Cisco Packet
Telephony Center


Monitoring and Troubleshooting

Cisco Packet Telephony Center (PTC) is a carrier-class network service management system for the packet-voice domain. Cisco PTC as the voice network service management application provides the foundation for fault, configuration, accounting, performance, and security (FCAPS) capability, enabling service providers to deliver a comprehensive end-to-end management solution. There are two products in the Cisco PTC line of products.

  • Cisco Packet Telephony Center-Virtual Switch (PTC-VS)—Cisco PTC-VS manages the media gateway controller and the media gateway combination of network elements as a single-managed object called the virtual switch.
  • Cisco Packet Telephony Center-Monitoring and Troubleshooting (PTC-MT)—Cisco PTC-MT is a strategic non-intrusive real-time network packet capture, call tracing, and call control analysis tool to help customers quickly and efficiently deploy, monitor, and troubleshoot voice over IP services on a data network.

Cisco Packet Telephony Center-Monitoring and Troubleshooting

Cisco PTC-MT is the ideal application to help identify inconsistencies among a large number of voice network elements during initial device staging and subsequently when all the devices are assimilated. In addition, Cisco PTC-MT offers valuable insight into network performance, real-time statistical reporting, and real-time call tracing to troubleshoot voice networks.

Because most problems are revealed during initial deployment or implementation of new VoIP services, Cisco PTC-MT is designed to help expedite the detection, isolation, and identification of these problems, thereby accelerating VoIP service deployment.

Cisco PTC-MT is a powerful, non-intrusive tool that monitors and correlates voice signaling and control events, including exceptions and error conditions in a packet telephony network. This includes capturing, decoding, and correlating events from Signaling System 7 (SS7) (see Figure 2 and Table 1) and VoIP signaling control protocols.

The Challenges to Troubleshooting

The call signaling messages in a VoIP network are distributed amongst different network elements for example like call agent, media gateway, gatekeeper, and route servers just to name a few network devices. To troubleshoot call control problems, network administrator needs access to the call signaling message from each of these network elements. It is time consuming to collect these signaling messages manually from each of network element. Once the data is collected, the administrator is still required to visually inspect and sort through many messages to locate relevant and critical diagnostic information. The volume of data collected and the VoIP expertise needed to analysis the information is both time consuming and a challenging task for any network administrator without the aid of a monitoring and troubleshooting tool like Cisco PTC-MT.

The Benefits of PTC-MT

With Cisco PTC-MT, the network administrator leverages the application to remove the uncertainty involved in detecting, isolating, and identifying voice-signaling problems among the numerous network elements in the VoIP network. Cisco PTC-MT automatically translates cryptic binary code to human readable formats. It then correlates these different signaling control protocols and their linkage to equivalent packet protocols based on a unique identifier for each call. It then presents the information to the user in a convenient and intuitive GUI that uses standard Internet Web browsers. Figure 1 shows an example of the Cisco PTC-MT GUI.

The many features and display filter options of PTC-MT help the user to view the network as a whole, isolate problem areas, and then analyze to identify detailed information for troubleshooting. For example, users can view key information in a condensed format—one line per call, or view detailed information with several lines per call.


Figure 1
Example of Some Available Options Within the Cisco PTC-MT Graphical User Interface

Key Features and Benefits of the Cisco PTC-MT

  • Automatic call-leg correlation for end-to-end call tracing
  • Real-time operation, including discovery and monitoring of signaling traffic to provide real-time feedback of network conditions
  • Non-intrusive monitoring that provides accurate network information without network performance degradation or network service interruption
  • Reports for abnormally disconnected calls for exception monitoring
  • Statistics reports that provide a quick view of real-time network performance trends like Answer Seizure Ratio (ASR) and Network Efficiency Ratio (NER) refer to Figure 2 for an example of SS7 for statistics report and Table 2 for an example of SIP statistic report.

Protocols Supported by Cisco PTC-MT

  • Session Initiation Protocol (SIP)
  • Media Gateway Control Protocol (MGCP)
  • H323
  • Signaling Link Terminal (SLT) Backhaul
    • SS7 ISDN User Part (ISUP)
    • ANSI/ITU/Japan/Singapore/Taiwan—China TUP—BTNUP
  • Primary Rate Interface (PRI) Backhaul
    • CORSAIR/Q.931
    • ISDN
  • Session Manager (SM)

SM messages are encapsulated in:

  • RUDP messages. (SM)
  • Reliable User Datagram Protocol (RUDP)
  • National ISDN 2 (NI2)
  • NI2+ Maintenance Messages (NMM)
  • Redundant Link Manager (RLM)
  • Extended ISDN User Part (EISUP)
  • Gatekeeper Transaction Message Protocol (GKTMP)
  • Remote Authentication Dial-In User Service (RADIUS)
  • Skinny Client Control Protocol (SCCP)
  • ISUP over M3UA
  • H.245
  • H.225 segmentation

Cisco PTC-MT follows the IP telephony signaling even as it moves across the IP network and switches from one protocol to a different protocol.

The following correlations are supported:

  • SS7-MGCP
  • SIP-MGCP
  • SIP-MGCP-SS7
  • SCCP-MGCP/CAS
  • SCCP-PRI BACKHAUL
  • SCCP-H323
  • H323-RADIUS-GKTMP

Cisco PTC-MT Summary

Cisco PTC-MT is a carrier-class application designed to handle large networks and therefore supports several simultaneous users. The application is multithreaded and performance scales with the number of processors in the server. Real-time packets capture, protocol decoding, and call correlation helps to reduce the time needed to identify and locate call control signaling problem. This is a critical first step in addressing and resolving a voice-related problem.

Cisco PTC-VS and PTC-MT play an integral role in helping service providers realize reduced network operations and management costs, reduced time to market for new customer services, and higher revenue generation. The Cisco PTC product line contributes greatly to increased voice service revenue and a reduction in the providers' total cost of network ownership.

Cisco PTC-MT Statistic Report

Cisco PTC-MT provides the following SS7 and SIP performance diagnostic statistic reports.

Table 1   SS7 Statistics Report Fields

Field  Description 

IAM

Number and rate per second for initial address messages (IAMs) received to request ISUP or TUP to reserve an idle trunk circuit from the originating switch to the destination switch

ACM

Number and rate of address completion messages (ACMs) received for ISUP or TUP. ACM messages are sent by the destination switch to the originating switch to indicate that the remote end of the trunk circuit has been reserved.

ANM

Number and rate of answer messages (ANMs) for TUP. The destination switch sends an ANM to the originating switch after the called party picks up the phone, to indicate that the call has been answered.

REL

Number of release messages (RELs) for ISUP. If the caller hangs up first, the originating switch sends an ISUP REL to release the trunk circuit between the switches. If the called party hangs up first, or if the line is busy, the destination switch sends an REL to the originating switch indicating the cause of the release; for example, normal release or busy.

RLC

Number of release complete messages (RLCs) for ISUP. When the destination switch receives the REL, it disconnects the trunk from the called party's line, sets the trunk state to idle, and transmits an ISUP RLC to the originating switch to acknowledge the release of the remote end of the trunk circuit. When the originating switch receives (or generates) the RLC, it terminates the billing cycle and sets the trunk state to idle in preparation for the next call.

CLASS 0: NORMAL EVENT

The number and rate of the following types of Class 0 Normal events:

  • Unallocated nb(1)—Unallocated (unsassgned) number. The called party cannot be reached because the called number is not currently allocated.

CLASS 1: NORMAL EVENT

The number and rate of the following types of Class 0 Normal events:

  • Normal REL(16)—Sent when either the caller or called party "hangs up" the call (cause = 16)
  • Busy(17)—Sent by the destination switch when the called party line is busy (cause = 17)
  • No user resp(18)—Sent by the destination switch when the called party line does not respond (cause = 18)
  • No answ from user(19)—Sent by the destination switch when the called party line is not answered (cause = 19)
  • Normal REL Unspec(31)—Number of REL messages for an unspecified cause (cause = 17)

CLASS 2: RESOURCE UNAVAIL

The number and rate of resource unavailable events.

CLASS 4: SERV OPT

Not implemented.

CLASS 5: INVALID MSG

The number and rate of invalid messages.

CLASS 6: PROTOCOL ERROR

The number and rate of recovery timer expiration protocol errors-Recov timer exp(102). These errors indicate that a procedure has been initiated by the expiration of a timer in association with error handling procedures.

CLASS 7: INTERWORKING

The number and rate of interworking events.

ASR

The Answer seizure ratio (ASR) for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field.

NER

Network efficiency ratio (NER) for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field.


Figure 2
SIP Statistics


Figure 3
SIP Statistics

Table 2   SIP Statistics Report Fields

Field   Description  

INVITE

The number and rate of SIP INVITE messages for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field. The INVITE request asks a called party in a SIP session to join a conference call or establish a two-party call, and includes information about the session that the called party can interpret to decide whether it has the processing capability to join the session.

BYE

The number and rate of SIP BYE requests for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field; the BYE request is issued by either the caller or called party to indicate to the server that it wishes to release the call.

CANCEL

The number and rate of SIP CANCEL requests for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field; the CANCEL request cancels a pending request with the same Call-ID, To, From, and CSeq (sequence number only) header field values, but does not affect a completed request.

INFORMATIONAL 1xx

The number and rate of SIP INFORMATIONAL 1xx status code messages for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field.

INFORMATIONAL 1xx messages indicate that the server or proxy contacted is performing some further action and does not yet have a definitive response; a server should send a 1xx response if it expects to take more than 200 ms to obtain a final response.

The statistics display separates the INFORMATIONAL statistics into counters for the following informational message types:

  • Trying (100)—Some unspecified action is being taken on behalf of this call (for example, a database is being consulted), but the user has not yet been located.
  • Ringing (180)—The called user agent has located a possible location where the user has registered recently and is trying to alert the user.
  • Session progress (183)—A message indicates the session progress.

SUCCESSFUL 2xx

The number and rate of SIP SUCCESSFUL 2xx status code messages for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field;

The SUCCESSFUL 2xx status code indicates that a request was successful.

REDIRECTION 3xx

The number and rate of SIP REDIRECTION 3xx responses for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field;

REDIRECTION 3xx responses give information about the user's new location, or about alternative services that might be able to satisfy the call;

The REDIRECTION 3xx counters only tabulate Moved Temporarily (302) response codes. Moved Temporarily (302) response codes can indicate the duration of the redirection through an Expires header.

REQUEST FAILURE 4xx l

The number and rate of SIP REQUEST FAILURE 4xx responses for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field;

REQUEST FAILURE 4xx responses are definite failure responses from a particular server; the SIP statistics display separates the REQUEST FAILURE 4xx responses into values for the following types of request failures:

  • Unauthorized (401)—The request requires user authentication.
  • Proxy authent req (407)—The client must first authenticate itself with the proxy.
  • Request Timeout (408)—The SIP request timed out.
  • Gone (410)—The requested resource is no longer available at the server and no forwarding address is known. This condition is considered permanent.
  • Temporarily unav (480)—The called party's end system was contacted successfully but the called party is currently unavailable (for example, not logged in or the manner of login precludes communication with the called party).
  • Busy (486)—The called party's end system was contacted successfully but the called party is currently not willing or able to take additional calls.
  • Request Cancelled (487)—The request was canceled.

SERVER FAILURE 5xx

The number and rate of SIP SERVER FAILURE 5xx responses for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field;

SERVER FAILURE 5xx responses are failure responses given when a server has erred.

GLOBAL FAILURlES 6xx

The number and rate of SIP GLOBAL FAILURES 6xx responses for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field;

GLOBAL FAILURES 6xx responses indicate that a server has definitive information about a particular user and that the failure is permanent.

ASR

The ASR for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field.

NER

NER for the cumulative time of the real-time packet-capture-mode session or for the time period specified in the Measurement Time Period field.

Recommended Minimum Server System Requirements

Sun enterprise platforms or the Netra NEBS certified platforms with Solaris 8

  • 1 CPU, 440 MHz or faster
  • 1 GB RAM
  • 2 GB of swap space
  • 18 GB fixed disk storage

Cisco PTC-MT operates on various hardware platforms. For current Solaris-based network-management hardware requirements, please see the Sun Cisco Optimized Platform Recommended Part Numbers.

Recommended Minimum Client System Requirements

Online Product Information

For more information about Cisco PTC-VS see:

http:// www.cisco.com/en/US/products/sw/netmgtsw/ps2025/ index.html

For more information about Cisco PTC-MT see:

http://www.cisco.com/en/US/products/sw/netmgtsw/ps4883/ index.html