CDR provides the following data organized into columns:
• ID: Unique autoincrement identification of call. It is created on SQL INSERT.
• Datetime: Start of a call.
• Duration: Total length of a call from start to end.
• Codec: Audio codec used in a call.
• Caller num/name: Caller number and name from SIP header.
• SIP agent: Agent string from SIP header.
• Last response: Last SIP response, number and full text description.
• Caller/Called src RTP: Source IP address of incoming RTP packets
from caller or receiver.
• MOS: Mean Opinion Score.
• Delay distribution: Show variable delays.
• Loss distribution: Show loss packets distribution.
• Commands: Download WAV or PCAP files.
Jitter Monitoring
sipMON allows monitoring of
relevant jitter data for all calls.
sipMON uses jitterbuffer simulator to keep
both directions of calls synchronized.
Delay Monitoring
Show variable delays delimited by ‘:’.
First number is number of delays between
50-70ms, second is between 70-90,
next is 90-120, 120-150, 150-200,
200-300, 300-more.
Packets Transfer Monitoring
Show lost packets distribution delimited by ‘:’.
The first number counts loss of one
isolated packet. The second is two consecutive
lost packets, next is 3, 4, 5, 6, 7, 8, 9
and 10-infinite lost packets.
MOS Score
Mean Opinion Score.
There are three MOS score values: Fixed 50|
Fixed 200|Adaptive 500.
• Fixed 50: Simulated jitterbuffer for devices
with almost no jitterbuffer (max 50ms)
• Fixed 200: Simulated jitterbuffer for devices
with 200ms fixed jitterbuffer
• Adaptive 500: Simulated jitterbuffer for
devices with adaptive 500ms jitterbuffer
RTP Monitoring
sipMON displays a diagram of RTP stream from all
IP addresses, caller and call receivers.
RTP stream diagrams are separated
for both sources.
Live Calls
Real-time monitoring of ongoing calls.
This feature is still in beta and requires the latest sipMON with enabled TCP manager port.
Call Recording
sipMON automatically records all calls
estabilished over the users' PBXware.
sipMON can also decode speech and play it
over the sipMON GUI or save it to
the disk as WAV.
Supported codecs are G.711 alaw/ulaw and commercial plugins supports
G.729a/G.723/iLBC/ Speex/GSM.
Data Transfer
Call data is automatically saved to the pcap file
with either only SIP protocol or
SIP/RTP/RTCP protocols.
Files may be exported to the hard drive
at any moment.
Calls with all relevant statistics are saved
to internal sipMON database.