X-LITE

Based on the advanced architecture of eyeBeam 1.5 (CounterPath's carrier grade telephony client), the new X-Lite is designed to showcase some of the feature rich capabilities available with our commercial softphone such as superior audio and video quality, zero-touch configuration, IM & Presence, and a comprehensive personal address book.

Note: To install, you will require the following information: 
username, password, authorization name, domain or IP address.

X-Lite is a fully functioning softphone featuring many of the popular features of eyeBeam. With your existing connection to a VoIP or broadband service or an IP-PBX (e.g. Asterisk, enterprise LAN), it allows you to make calls to any softphone, mobile or landline number.

SJphone è uno dei software VoIP più noti ed utilizzati, in particolare è spesso offerto in combinazione con i servizi dei maggiori operatori VoIP. Il programma gira praticamente su qualsiasi cosa assomigli ad un PC: sono infatti supportati i sistemi Microsoft Windows 98SE/ME/2K/XP/2003, Linux e Mac OS X ed i PDA dotati di PocketPC 2002/2003/2003SE, Windows CE.NET 4.0/4.2/5.0, Windows Mobile 5.0 (il softphone è ottimizzato per l’utilizzo con Intel XScale® CPU).
SJphone è compatibile con i due principali standard di segnalazione delle chiamate, SIP e H.323 e funziona quindi con moltissimi operatori VoIP e sistemi IP-PBX. E’ possibile configurare diversi “servizi” (solitamente legati alla scelta di un operatore) e passare in modo semplice da uno all’altro. I codec disponibili per la trasmissione del segnale vocale sono G.711 A/u, GSM, Speex e iLBC.

MySpeed PC VoIP Standard simulates VoIP traffic over your Internet connection and provides an analysis of the voice quality, along with a report of the consistency of bandwidth provided by your ISP.

The key measures of connection quality for VoIP are not broadband speeds as many would think, but Jitter and Packet Loss. When Jitter or Packet Loss are high, it often results in a garbled communication similar to a bad cell phone connection. MySpeed PC VoIP measures both jitter and packet loss, and rates your connection for voice quality.

• Test your connection for VoIP sound quality.
• Automatically test your connection speeds at regular intervals, view your results over time
• See the number of VoIP lines supported by your Internet connection
• View graphical reports of the data transfers for detailed analysis to help identify the cause of poor performance
• Measure the consistency of your download speeds, a critical measure for time-sensitive applications such as VoIP

• Measures Jitter and Packet Loss
  The primary measures of VoIP connection quality are Jitter and Packet Loss. MyVoIPSpeed simulates voice traffic on your Internet connection, determines the levels of Jitter and Packet Loss, and rates the corresponding voice sound 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 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.
  
  The VoIP graph shows the variance of UDP jitter during the simulation test. This variance must be kept to a minimum otherwise call quality will be degraded. The packet loss distribution is shown in red, high packet loss will result in broken sound during calls.