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VoIP is an acronym for Voice over Internet Protocol (Voice over IP). It allows users to make free or cheap phone calls through broadband connection over the internet. The voice data is transferred over the Internet by means of digital packets.

VoIP is a revolutionary advance in our modern life. Nowadays, more and more people use VoIP to make phone calls (including international calls) with very low charge (toll). This technology has the potential to change the world¡¦s telecommunication companies and change our lifestyles.


Triple Play service is a term used to describe the delivery of VoIP (Voice over IP), IPTV, and Internet Access (Data) to consumers through a single broadband connection. Triple Play service has been a focal point on an integrated business model for the convenience of modern lifestyle.

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VoIP Expert Analysis software employs MOS ITU-T G.107 and R-Factor to evaluate and rate the quality of telephony voice traffic, incorporate an extended version of the E Model analyzing the effects of time varying IP network impairments which provides a more accurate of user option. The audio quality testing application¡¦s flexibility and user-friendliness enable you to analyze the call quality in many ways such as non-intrusive at Ethernet interface or intrusive at DSL analog loop. It allows you to set up the analysis using 3 parameters, chosen between R-factor to MOS (Mean Option Score), Listener Quality (MOS-LQ), MOS PESQ (MOS-PQ), Conversational Quality (MOS-CQ), G.107 and R Conversation and evaluate those against each other.
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> VoIP Key Features

Triple Play Analysis Suite - VoIP (Expert Analysis)
Non-intrusive voice quality monitoring and evaluation based on the
  industry standard ITU-T G.107 E Model (MOS and R-Factor).
Non-intrusive SIP video phone quality monitoring and evaluation based
  on the patented Telchemy live call method to provide a real-time
  assessment of video quality in term of Video Mean Opinion Score
Support SIP based signaling analysis per RFC 3261 in Ladder Diagram,
  every SIP call/transaction is individually recorded and traced in graphic
  Call Flow to present signaling point registered with IP address/port
  number, delta time and an alarm icon indicating the violation of SIP
RTP voice stream analysis in real time including RTP Jitter, Packet
  loss, Codec Type, SSRC identify and Session base throughput¡Ketc.,
  per industry standard RFC 1889.
Option applicable for CSA¡¦s ADSL, VDSL and GigE Platform Interface
  Modules (PIM).
Events logged with time stamping.
Triple Play Analysis Suite - VoIP (VoIP Simulation)
Active Call with Real-Time Information & Phone Book function.
Dedicated Function Keypads (Dial, Hand Up, Hold, and Transfer).
Support for audio codecs: G.711u, GSM, G.711a, and G.729a.
STUN support and RFC 2833 Out-Of-Band DTMF tones.
Support Network Client simulation including MAC spoofing, PPPoE, Fix
  IP and DHCP with VCI(option 60) and UCI(option 77) configurable.
MOS vs. R-Factor
Measuring call quality provides objective and subjective scores to evaluate existing call conditions and compare them with historical conditions. R-factor and MOS are two well-known methods for measuring call quality and overall VoIP network health.
A MOS can range from 5 down to 1. An estimate of the MOS can be directly calculated from the R factor. The R-factor uses a scale of between 100 and 1 to measure call quality.

However, call quality are influenced by equipment impairments, audio codec, Jitter buffer overruns, and packet loss. Thus, the maximum R factor is about 94 (the highest MOS is 4.4) because of the degradation during converting an actual voice conversation to a network signal.

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