Enable End-Of-Call Statistics Reports In Sip Messages - Cisco 8800 Series Manual

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Enable End-of-Call Statistics Reports in SIP Messages

Enable End-of-Call Statistics Reports in SIP Messages
You can enable the phone to send end-of-call statistics in Session Initiation Protocol (SIP) messages (BYE
and re-INVITE messages). The phone sends call statistics to the other party of the call when the call terminates
or when the call is on hold. The statistics include:
• Real-time Transport Protocol (RTP) packets sent or received
• Total bytes sent or received
• Total number of lost packets
• Delay jitter
• Round-trip delay
• Call duration
The call statistics are sent as headers in SIP BYE messages and SIP BYE response messages (200 OK and
re-INVITE during hold). For audio sessions, the headers are
the headers are
Example of call statistics in a SIP BYE message:
Rtp-Rxstat: Dur=13,Pkt=408,Oct=97680,LatePkt=8,LostPkt=0,AvgJit=0,VQMetrics="CCR=0.0017;
ICR=0.0000;ICRmx=0.0077;CS=2;SCS=0;VoRxCodec=PCMU;CID=4;VoPktSizeMs=30;VoPktLost=0;
VoPktDis=1;VoOneWayDelayMs=281;maxJitter=12;MOScq=4.21;MOSlq=3.52;network=ethernet;
hwType=CP-8865;rtpBitrate=60110;rtcpBitrate=0"
Rtp-Txstat: Dur=13,Pkt=417,Oct=100080,tvqMetrics="TxCodec=PCMU;rtpbitrate=61587;rtcpbitrate=0
Rtp-Videorxstat: Dur=12;pkt=5172;oct=3476480;lostpkt=5;avgjit=17;rtt=0;
ciscorxvm="RxCodec=H264 BP0;RxBw=2339;RxReso=1280x720;RxFrameRate=31;
RxFramesLost=5;rtpBitRate=2317653;rtcpBitrate=0"
Rtp-Videotxstat: Dur=12;pkt=5303;oct=3567031;ciscotxvm="TxCodec=H264 BP0;TxBw=2331;
TxReso=1280x720;TxFrameRate=31;rtpBitrate=2378020;rtcpBitrate=0"
For description of the attributes in call statistics, see
217.
You can also use the
<Call_Statistics ua="na">Yes</Call_Statistics>
Before you begin
Access the phone administration web page, see
Procedure
Step 1
Select Voice > SIP.
Step 2
In the RTP Parameters section, set the Call Statistics field to Yes to enable the phone to send call statistics
in SIP BYE and re-INVITE messages.
Cisco IP Phone 8800 Series Multiplatform Phone Administration Guide for Release 11.3(1) and Later
216
and
RTP-VideoRxStat
RTP-VideoTxStat
parameter in the phone configuration file to enable this feature.
Call_Statistics
and
RTP-RxStat
.
Attributes for Call Statistics in SIP Messages, on page
Access the Phone Web Interface, on page
Cisco IP Phone Configuration
. For video sessions,
RTP-TxStat
104.

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