We are SAIB bank in Saudi Arabia country, looking to monitor cisco IVR system using DCRUM 12.4 .
Is there any best practices to configure and monitor this system using DCRUM 12.4?
What's the value i can get and if there some existing report samples ?
Specifically what are you looking to measure with DC RUM within this environment? are you looking at monitoring CTI/call control data? or the actual VoIP performance of the end users? DC RUM has no formal interface to the IVR systems, the exchanges are made up of many protocols and basic performance and availability metrics might be visible, depending on where you place the AMD.
The VoIP decode was specifically designed to produce performance metrics related to the end users , it will not report on any specific script or service message exchanged between the IVR system and IP phones/desktop agents, but will potentially provide basic call metrics in terms of MOS, latency , jitter etc between the two systems, this can then be supplemented by looking at the IP phone/desktop agent to end user/gateway connection, in order to see specific VoIP degradation and issues on the associated legs, as such it is best to place the AMD in a position to capture and see that exchange from the IP phones/desktop agents (ingress/eghress postion) as opposed to the IVR system.
Thanks Mike for Reply,
Actually we need to monitor both, but currently the bank looking to monitor end user specially when the customer used IVR service and complaining.
The cisco IVR system used SIP protocol and XML service in between to handle the customer service.
Appreciated your help
The XML should be visible with the DC RUM decodes.But in terms of SIP you have to remember that whilst we have a SIP decode in DC RUM, and you should potentailly get basic avaiblity, it will not show all the status codes, it is instead designed specifically for the end user VoIP, meaning that it is primarily used to a associated the RTP audio streams, and as a result the status codes that are observed are recorded under general headings such as calls not started due to remote peer, calls finished with termination error etc. In addition the SIP decode doesn't support forking, so calls diverted by the IVR won;t necessarily be shown as a single contiguous call.
A Customer demands monitoring some specific sip status code. What are the SIP Status Codes (and H.323 Cause Codes if possible), monitored by each of 3 error metrics at VOIP decode (12.4 and 2017/2018 if changed)?
Obs.: I doesn't found any more details at documentation / community.