True Business-Class Reliability makes VONx Voice over Internet Protocol (VoIP) service the premiere quality-of-service VoIP offering in the mid-South. And, VonX managed-network service is available, now.
VoIP over the public Internet suffers from effects like latency, packet loss and jitter. ISDN-Net's VonX improves upon VoIP by routing packets over a dedicated, constantly monitored DSL/T1 line so that the packets never touch the public Internet. Some of the factors that affect ordinary VoIP as carried over the Internet are described below. See "The VonX Advantage" to find out the advantage of ISDN-Net's approach to VoIP.
Data networks like the Internet were designed to carry data like text files. When files are transmitted over the Internet, they are broken up into packets that can take various routes (some of which are slower than others) before arriving at their destination. They may arrive out of sequence and have to be reassembled in order. There may be delays between one packet and the next. This isn't a big concern with ordinary files, but it can drastically affect a voice conversation, perhaps rendering parts of it unintelligible at times.
A two-way phone conversation is sensitive to latency, the time it takes for a packet to travel from the sender to the receiver. Most people notice round-trip latency of about one-quarter second, and as latency increases conversation becomes correspondingly difficult, even disconcerting. Several things contribute to latency, including
- Backbone (network) latency. These are the normal delays encountered when forwarding a packet from point to point in a network.
- Codec latency. Codec stands for Coder/Decoder, which is the conversion of analog sound to digital data and vice-versa. Codec latency is the time it takes to convert the voice signal to data packets and back to sound.
- Jitter buffer size. Packets containing the voice data can arrive at the destination at slightly different times, so to smooth out the packet flow they must be kept in a kind of "bucket" before they are converted back to an analog voice signal. Unfortunately, the larger the buffer, the worse the latency, so a compromise is usually adopted.
- Total network load. Much like automobile traffic, the more data traffic flowing through a network, the greater the likelihood of a delay.
- Packet loss. Packets occasionally get lost on the Internet. For ordinary data, this can be remedied simply by resending the missing packet. This won't work for voice, of course, because if a packet is missing it would take too much time to resend it.
- Call setup time. When you pick up your telephone handset, you expect an immediate dial tone. And when you dial the number, you expect to hear ringing from the other end in short order. You'll notice if the delays are excessive.
- Call success ratio. You naturally expect that when you dial the phone, you're going to reliably connect to person at the other end; you expect a high call success ratio from your VoIP connection.