Call 1 Status/Call 2 Status - Cisco 8831 Administration Manual

Info
Parameter
Next Registration In
Mapped SIP Port

Call 1 Status/Call 2 Status

The following parameters show for each line and call on the phone.
Parameter
Call State
Duration
Remote Address
Local Address
Start Time
Type
Peer Name
Peer Phone
Sender Packets
Sender Octets
Sender Codec
Rcvr Lost Packets
Avg Jitter
Rcvr Codec
Rcvr Packets
Rcvr Octets
Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide
A-4
Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference
Description
Number of seconds before the next registration renewal.
Port number of the SIP port mapped by NAT.
Description
Status of the call.
Duration of the call.
Address of the remote device.
Address of the local device.
Starting time of the call
Direction of the call.
Name of the internal phone.
Phone number of the internal phone.
Number of RTP voice packets transmitted since voice stream
was opened.
This number is not necessarily identical to the number
Note
of RTP voice packets transmitted since the call began
because the call might have been placed on hold.
Total number of octets sent by the phone.
Type of voice stream transmitted (RTP streaming audio from
codec): G.729, iLBC, G.711 u-law, or G.711 A-law.
Missing RTP packets (lost in transit.)
Estimated average RTP packet jitter (dynamic delay that a
packet encounters when going through the network) observed
since the receiving voice stream was opened.
Type of voice stream received (RTP streaming audio from
codec): G.729, iLBC, G.711 u-law, or G.711 A-law.
Number of RTP voice packets received since voice stream was
opened.
This number is not necessarily identical to the number
Note
of RTP voice packets received since the call began
because the call might have been placed on hold
Total number of octets received by the phone.