Cisco CP-7937G Administration Guide - Page 78
Call Statistics Screen, Applications, Settings, Status, Call Statistics
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Status Menu Chapter 7 Viewing Model Information, Status, and Statistics on the Conference Station Call Statistics Screen The Call Statistics screen displays information about the last call on the conference station. Table 7-2 describes the information displayed on the screen. Note You can remotely view the call statistics information by using a web browser to access the Streaming Statistics web page. For more information about remote monitoring, see Chapter 8, "Monitoring the Conference Station Remotely." A single call can have multiple voice streams, but data is captured for only the last voice stream. A voice stream is a packet stream between two endpoints. If one endpoint is put on hold, the voice stream stops even though the call is still connected. When the call resumes, a new voice packet stream begins, and the new call data overwrites the former call data. To display the Call Statistics screen for information about the last voice stream, choose Applications > Settings > Status > Call Statistics. To exit the Call Statistics screen, press Exit. Table 7-2 Call Statistics Items Item Remote Address Local Address Start Time Codec Type Payload Size Rcvr Packets Rcvr Lost Packets Rcvr Octets Rx Expected Pkts Last Rx Seq No Most recent Rx SSRC Avg Jitter Max Jitter Description IP address and UDP port of the stream. IP address and UDP port of the conference station. Internal time stamp indicating when Cisco Unified Communications Manager 6.0 requested that the conference station start transmitting packets. Type of voice stream received or transmitted (RTP streaming audio): G.729, G.711 u-law, G.711 A-law, G.722, G.722.1, or Lin16k. Size of voice packets, in milliseconds, in the receiving or transmitting voice stream (RTP streaming audio). 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. Missing RTP packets (lost in transit). Number of bytes of voice packets received since voice stream was opened. The expected number of packets received for the local conference station. The sequence number of the last RTP packet received. The Synchronization Source field of the last RTP packet received. Estimated average RTP packet jitter (dynamic delay that a packet encounters when going through the network) observed since the receiving voice stream was opened. Maximum jitter observed since the receiving voice stream was opened. Cisco Unified IP Conference Station 7937G Administration Guide for Cisco Unified Communications Manager 6.0 7-4 OL-11560-01 Rev. B0