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WEB PAGE CUSTOMIZATION
The MyVoIPSpeed Server speed test applet can be easily incorporated into your own web pages. Details.


LANGUAGES
The MyVoIPSpeed Server display text can be easily translated to another language, just edit the text in the myspeed.bin file to change the display.


PERFORMANCE
DATABASE
The Professional and NOC Editions record all speed test results in a server-side database that provides visibility to end-user bandwidth performance.

The Session ID option enables easy tagging of results to identify bandwidth performance from individual customers, web pages, locations, etc.



MuVoIPSPEED
FOR YOUR PC
Looking for an accurate VoIP & speed test program to run bandwidth tests from your PC?

MyVoIPSpeed PC
Basic VoIP simulation and speed testing

MyVoIPSpeed PC
Advanced

Adds automated test options, performance database, flexible reporting, and more!

 


MyVoIPSpeed Server simulates a VoIP connection and measures the key factors affecting connection quality. The information below is provided in the VoIP test results.
    test voip, voip test, voip performance
MYVOIPSPEED SERVER FEATURES
VoIP ANALYSIS
VoIP Simulation Test
• All editions
UDP VoIP traffic is simulated between your server and the browser client, diagnostic tests are performed to measure the critical factors that impact VoIP connection quality.
MOS Score
• All Editions
MOS is a numeric measure of the sound quality of a communication connection. Although the score is subjective it provides a widely-used method to rate the quality of voice communication.

The score is normally between 1 and 5 with 5 being the best. In MyVoIPSpeed, a VoIP simulation that drops below 3.5 is considered poor quality, a measure of 4.2-4.5 is considered good quality.
Jitter
• All Editions
Jitter is a key factor in assessing VoIP connection quality.

Jitter is the variation in time between packets sent and packets arriving caused by network difficulties such as route changes, congestion, packet loss, traffic regulators etc., and plays a major role in the quality of a VoIP call. VoIP works by sending voice data as a stream of packets from source to destination. These packets can take a varying amount of time to reach the destination and invariably do not arrive in the order in which they were sent.

For a VoIP telephone call to work well the packets sent from the source must arrive within a certain time window (or ‘buffer’) in order for the receiving end to reassemble the packets in the correct order and reproduce the spoken words. When there is excessive jitter the time delay is too long (high latency) and packets arrive outside the time window and get lost from the call, or ‘discarded’. As a result, the recomposed sound no longer reflects exactly what was sent, and depending of the extent of the delay may not be understandable by the recipient.
Packet Loss
• All Editions
Packet loss plays a key role in the quality of VoIP connections, as high packet loss causes some of the voice data not to arrive to the recipient. Packet loss occurs when voice packets are discarded by the jitter buffer, or dropped by network routers/switches due to high congestion.

MyVoIPSpeed measures the percentage of packet loss and reports the associated level of sound quality. For more accurate results Packet Loss is measured in one direction only, which is normally not possible using traditional methods such as ping, where routing can affect the reply.
TraceRoute Analysis
• NOC Edition
The NOC Edition includes RTA (remote test agents) that can be deployed to multiple remote systems for automated VoIP and bandwidth testing. The RTAs can run traceroute tests and report the diagnostic information to the central performance database. .

Internet connections involve multiple networks and device, and the routing path between 2 points can change frequently, complicating the process of pinpointing connectivity problems. Traceoute analysis is extremely useful in identifying the network device or segment that is causing poor performance.
Advanced Statistics
• Professional & NOC Editions
The Advanced Statistics report tab provides a summary of the above test results for each VoIP simulation test, along with more detailed statistics including:
• Jitter: from server to client
• Jitter: from client to server
• Packet Loss: from client to server
• Packet Loss: from server to client
• Packets out of order
• Number of supported VoIP lines
BROADBAND SPEED TEST

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