GoTo Logo | Network Test

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Network tested for the following products:

Data Center Connection
Testing the closest data center to ensure the quickest connection and best possible call quality.
New York
Chicago
Atlanta
Los Angeles
Dallas
Las Vegas
São Paulo
Frankfurt
Sydney

Packet Loss

Occurs when packets fail to reach their destination. A high rate of packet loss will affect call quality. We recommend a ratio less than 1 percent.

Latency

To ensure the best user experience, the end-to-end latency of a VOIP system should be no more than 150 ms (including voice encoding, transport and decoding). This test measures the network latency (transport), which should be less than 100 ms.

Jitter

Voice packets travelling on a network from point A to point B don't always take the same time to travel (and sometimes even arrive out of order). The jitter is a measure of the variance of packet latency. A jitter lower than 30 ms is recommended.

Bandwidth

This represents the amount of data your network can transmit/receive in a given amount of time. We measure downstream and upstream bandwidth (incoming and outgoing data) separately then simultaneously.

More bandwidth means more concurrent calls.

Bandwidth usage per concurrent calls

G.729 G.711/G.722
1 call 0.008 Mbps 0.064 Mbps
5 calls 0.040 Mbps 0.320 Mbps
10 calls 0.080 Mbps 0.640 Mbps
50 calls 0.400 Mbps 3.200 Mbps
100 calls 0.800 Mbps 6.400 Mbps

NOTE: Both simultaneous upstream and downstream bandwidth measures need to be greather than the targeted usage scenario.

SIP ALG

Stands for Application Layer Gateway, and is common in many commercial routers. This should be disabled to prevent the router from modifying SIP packets.

Open Ports

Test whether ports required for normal operations are blocked.

DNS SRV Lookup

GoToConnect's SIP servers are discovered using SRV lookups. Your DNS server needs to process these appropriately.

NAT Settings

NAT binding timeout is the length of time that the router still keeps an inactive UDP NAT session. We strongly recommend a delay greater than 30 sec.

Open Ports

Test whether ports required for normal operations are blocked.

Domain DNS Lookup

LogMeIn services are accessible using specific domain names. Your DNS server needs to resolve these appropriately.

Packet Loss

Occurs when packets fail to reach their destination. A high rate of packet loss will affect call quality. We recommend a ratio less than 1 percent.

Latency

To ensure the best user experience, the end-to-end latency of a VOIP system should be no more than 150 ms (including voice encoding, transport and decoding). This test measures the network latency (transport), which should be less than 100 ms.

Bandwidth

This represents the amount of data your network can transmit/receive in a given amount of time. We measure downstream and upstream bandwidth (incoming and outgoing data) separately then simultaneously.

More bandwidth means more concurrent calls.

Bandwidth usage per concurrent calls

Simulatenous meeting Audio WebCam Screen Sharing ALL (minimum)
1 0.04 Mbps 0.7 - 2 Mbps 0.7 - 8 Mbps 1 Mbps
5 0.2 Mbps 3.5 - 10 Mbps 3.5 - 40 Mbps 5 Mbps
10 0.4 Mbps 7 - 20 Mbps 7 - 80 Mbps 10 Mbps
20 0.8 Mbps 14 - 40 Mbps 14 - 80 Mbps 20 Mbps

NOTE: Both simultaneous upstream and downstream bandwidth measures need to be greather than the targeted usage scenario.

Open Ports

Test whether ports required for normal operations are blocked.

Domain DNS Lookup

LogMeIn services are accessible using specific domain names. Your DNS server needs to resolve these appropriately.

Packet Loss

Occurs when packets fail to reach their destination. A high rate of packet loss will affect call quality. We recommend a ratio less than 1 percent.

Latency

To ensure the best user experience, the end-to-end latency of a VOIP system should be no more than 150 ms (including voice encoding, transport and decoding). This test measures the network latency (transport), which should be less than 100 ms.

Bandwidth

This represents the amount of data your network can transmit/receive in a given amount of time. We measure downstream and upstream bandwidth (incoming and outgoing data) separately then simultaneously.

More bandwidth means more concurrent calls.

Bandwidth usage per concurrent calls

Simulatenous meeting Audio WebCam Screen Sharing ALL (minimum)
1 0.04 Mbps 0.7 - 2 Mbps 0.7 - 8 Mbps 1 Mbps
5 0.2 Mbps 3.5 - 10 Mbps 3.5 - 40 Mbps 5 Mbps
10 0.4 Mbps 7 - 20 Mbps 7 - 80 Mbps 10 Mbps
20 0.8 Mbps 14 - 40 Mbps 14 - 80 Mbps 20 Mbps

NOTE: Both simultaneous upstream and downstream bandwidth measures need to be greather than the targeted usage scenario.

SIP ALG

Stands for Application Layer Gateway, and is common in many commercial routers. This should be disabled to prevent the router from modifying SIP packets.

Local Network Information

Client IP Address:
Client MAC Address:
Default Gateway IP Address:
Default Gateway MAC Address: