Velocity Virtual Office - FAQ - How to interpret Remote Monitor results for a best effort connection

How to interpret Remote Monitor results for a best effort connection

The remote monitor sends four packets every second (one packet every 250 milliseconds) to one of our SIP proxies which mirrors the packets back to the remote monitor. All statistics reflect full round trip time (RTT). These are not one-directional statistics, but rather statistics for the trip up to our server and back to the host.

This document outlines the statistics in the remote monitor reports received every 5 minutes. Each of these reports contains five minutes worth of statistics.

The individual field descriptions are as follows:
Pay particular attention to "Pkts Short +200", "Pkt Loss %", and "SIP ALG Detection"

Date/Time The date and time of the report. Reports are sent every five minutes and contain a five minute snapshot of round trip statistics.
Public IP The IP address of the last public interface the packet was received from. Helps identify the location the remote monitor is running at.
Local IP The local IP address of the host machine the packets are coming from.
RTT Short The shortest round trip time of any of the packets sent during the five minute test.
RTT Long The longest round trip time of any of the packets sent during the five minute test.
RTT Avg The average of all of the round trip times during the five minute test.
RTT Std Dev The Standard Deviation calculation of all of the round trip times during the five minute test.
Pkts Short +100 The quantity of packets whose round trip time was at least 100 milliseconds greater than RTT Short.
Pkts Short +150 The quantity of packets whose round trip time was at least 150 milliseconds greater than RTT Short.
Pkts Short +200 The quantity of packets whose round trip time was at least 200 milliseconds greater than RTT Short.
This is the best way of determining jitter effects on a conversation from the statistics. Jitter is the change latency from one packet to the next in a VoIP stream. Jitter buffers on VoIP phones attempt to absorb changes in latency (jitter) by buffering packets; these buffers can typically absorb about 100-120ms of jitter at most before packets get dropped from the audio stream (or latency in the audio would suffer greatly). Changes in latency between packets that exceed what the jitter buffer can absorb will affect audio quality in the same way as packet loss.
Five or more packets consistently in the Pkts Short +200 column will begin to affect audio quality. Once 20 or more packets are reached audio quality may begin to approach 'cell phone quality' service. If they are close to 100 it is likely unusable for VoIP conversations due to the degrading effects on maintaining audio conversations; this represents consistent and severe congestion.
Pkts Sent The total number of packets sent which are included in this report. Note that the report may be just short of a full five minute sample since packets in process are not included in the statistics. This can be up to a two second window of packets that are ignored, or eight packets which do not make it into the report.
Pkts Recvd The number of packets received back during the five minute test.
Pkts Lost The number of packets lost on the network during the five minute test.
Pkt Loss % The percent of packets lost during the five minute test.
Packet loss of less than 0.5% is typically not audible.
Packet loss of 0.5% or greater starts to become audible; as it reaches 1% or above it may start to sound closer to ‘cell phone’ audio quality for conversations.
Packet loss of 1% or greater is always audible and may be reported as periodically ‘losing a word’ during conversations. Packet loss of 5% or above is essentially unusable for VoIP as the conversation starts to become unintelligible making it difficult or impossible to conduct a conversation.
SIP ALG Detection SIP ALGs are SIP (Session Initiation Protocol) Application Level Gateways, or special code built into routers that process and modify VoIP packets in an attempt to help traverse NAT (Network Address Translation).
Virtual Office NEVER needs the assistance of a SIP ALG to function. And in essentially every case we've encountered they will create periodic unreliability issues and/or have bugs which will adversely affect performance (cause one way audio, calls that do not ring, calls that do not connect, etc). This is due to their periodic improper handling of these packets over time as they trying to modify and watch the state of these VoIP packets to perform NAT. Virtual Office does not use nor need SIP ALGs in any way and they should always be disabled.
IN ALL CASES IF A SIP ALG IS DETECTED IT SHOULD BE DISABLED ON THE ROUTER IF AT ALL POSSIBLE TO AVOID THE SPORADIC AND UNPREDICTABLE RELIABILITY ISSUES WHICH CAN AND TYPICALLY ARE CAUSED BY THESE SIP ALGs.
Remote monitor versions prior to v1.0.008 do not have the SIP ALG detector built in and this column will be blank for these versions. To upgrade the remote monitor to include a SIP ALG detector uninstall, reboot, re-download, and re-install the RemoteMonitorSetup.exe to get the latest version. Or if you have a Java enabled browser you may use our Applet based SIP ALG detector to see if you have a SIP ALG functioning on your network.

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Date/Time	The date and time of the report. Reports are sent every five minutes and contain a five minute snapshot of round trip statistics.
Public IP	The IP address of the last public interface the packet was received from. Helps identify the location the remote monitor is running at.
Local IP	The local IP address of the host machine the packets are coming from.
RTT Short	The shortest round trip time of any of the packets sent during the five minute test.
RTT Long	The longest round trip time of any of the packets sent during the five minute test.
RTT Avg	The average of all of the round trip times during the five minute test.
RTT Std Dev	The Standard Deviation calculation of all of the round trip times during the five minute test.
Pkts Short +100	The quantity of packets whose round trip time was at least 100 milliseconds greater than RTT Short.
Pkts Short +150	The quantity of packets whose round trip time was at least 150 milliseconds greater than RTT Short.
Pkts Short +200	The quantity of packets whose round trip time was at least 200 milliseconds greater than RTT Short. This is the best way of determining jitter effects on a conversation from the statistics. Jitter is the change latency from one packet to the next in a VoIP stream. Jitter buffers on VoIP phones attempt to absorb changes in latency (jitter) by buffering packets; these buffers can typically absorb about 100-120ms of jitter at most before packets get dropped from the audio stream (or latency in the audio would suffer greatly). Changes in latency between packets that exceed what the jitter buffer can absorb will affect audio quality in the same way as packet loss. Five or more packets consistently in the Pkts Short +200 column will begin to affect audio quality. Once 20 or more packets are reached audio quality may begin to approach 'cell phone quality' service. If they are close to 100 it is likely unusable for VoIP conversations due to the degrading effects on maintaining audio conversations; this represents consistent and severe congestion.
Pkts Sent	The total number of packets sent which are included in this report. Note that the report may be just short of a full five minute sample since packets in process are not included in the statistics. This can be up to a two second window of packets that are ignored, or eight packets which do not make it into the report.
Pkts Recvd	The number of packets received back during the five minute test.
Pkts Lost	The number of packets lost on the network during the five minute test.
Pkt Loss %	The percent of packets lost during the five minute test. Packet loss of less than 0.5% is typically not audible. Packet loss of 0.5% or greater starts to become audible; as it reaches 1% or above it may start to sound closer to ‘cell phone’ audio quality for conversations. Packet loss of 1% or greater is always audible and may be reported as periodically ‘losing a word’ during conversations. Packet loss of 5% or above is essentially unusable for VoIP as the conversation starts to become unintelligible making it difficult or impossible to conduct a conversation.
SIP ALG Detection	SIP ALGs are SIP (Session Initiation Protocol) Application Level Gateways, or special code built into routers that process and modify VoIP packets in an attempt to help traverse NAT (Network Address Translation). *Virtual Office NEVER* needs the assistance of a SIP ALG to function.** And in essentially every case we've encountered they will create periodic unreliability issues and/or have bugs which will adversely affect performance (cause one way audio, calls that do not ring, calls that do not connect, etc). This is due to their periodic improper handling of these packets over time as they trying to modify and watch the state of these VoIP packets to perform NAT. Virtual Office does not use nor need SIP ALGs in any way and they should always be disabled. IN ALL CASES IF A SIP ALG IS DETECTED IT SHOULD BE DISABLED ON THE ROUTER IF AT ALL POSSIBLE TO AVOID THE SPORADIC AND UNPREDICTABLE RELIABILITY ISSUES WHICH CAN AND TYPICALLY ARE CAUSED BY THESE SIP ALGs. Remote monitor versions prior to v1.0.008 do not have the SIP ALG detector built in and this column will be blank for these versions. To upgrade the remote monitor to include a SIP ALG detector uninstall, reboot, re-download, and re-install the RemoteMonitorSetup.exe to get the latest version. Or if you have a Java enabled browser you may use our Applet based SIP ALG detector to see if you have a SIP ALG functioning on your network.