What are the Voice over IP Connection Quality Issues One Should Know

What are the Voice over IP Connection Quality Issues One Should Know

Whatever the reason for the loss of the packet – the receiving buffer, the local network or the global network – the result is one: this packet was not received by the addressee. Likewise, the delay introduced by the wireless LAN (WLAN) is added together with the latency of the wired LAN, WAN and wired LAN on the recipient’s side.

The article discusses the main elements that affect the quality of voice packet transmission. In particular, we will consider the existing requirements for delay, jitter and loss, as well as determine the performance of various network services that affect the transmission of voice packets. Fortunately, rich experience of operating local and global networks for the transmission of voice packets allows us to locate the problem with a high degree of confidence before the subscribers complain of a bad connection.

Quality issues in packet telephony

The use of packet technology is associated with three main problems of voice service: voice quality, delay and echo cancellation.

The quality of voice transmission. First of all, the quality of the signal in the voice connection is determined by the method of speech coding and the proportion of packets not reaching the receiver that are to be decoded. When transmitting voice over IP networks, packets may be lost for the following two reasons.

  1. Networks lose packets due to errors or buffer overflow.
  2. The RTP (Real-Time Transport Protocol) receive delay buffer can skip packets if they arrive at a delay greater than the one for which the buffer is calculated. Thus, the arrival of a package with a delay is tantamount to its absence.

The way that the packet loss affects the transmission of a signal depends on the method of voice coding. The question of acceptable packet loss, as well as other problems of voice coding, will be considered below.

Transmission delay. This is the total delay of the voice signal when it is transmitted. It is determined by a number of factors related to the features of the local and global network. These include voice encoding/compression, packet generation, channel conflict (in WLAN), network transport/network buffering, and jitter elimination. It is necessary to know that if the delay in one-way communication exceeds 150 ms, it begins to affect the rate of conversation. Distance, router buffering and conflicts in WLANs are factors that affect the total transmission delay. This indicator is among those basic factors, the quality of which is censured in packet telephony.

Another problem associated with signal transmission time is jitter or a change in the packet-to-packet delay value caused by dynamic buffering in the packet network.

If this effect is not eliminated, the voice becomes illegible. The RTP protocol helps to eliminate jitter, however, during the use of RTP, an additional transmission delay is accumulated.

Control the echo. In all telephone networks, there is an echo effect. However, if the delay in one-way transmission exceeds 35 … 40 ms, this effect becomes very noticeable and annoying. If the delay exceeds the value of this parameter, you must use the echo cancellation equipment. In fact, in all networks of voice packet transmission, the delay in one direction exceeds 40 ms, therefore, in the development of the system, an echo control mechanism should be provided.

Voice Quality

The human voice is analogous in nature. When we say, the vocal cords generate vibrations, representing a certain sequence of compressions and discharges of the air medium, i.e. Analog (continuously changing) signal. Before passing it over a packet network with digital data, a codec (encoder and decoder) is required to convert this signal into a digital representation.

When choosing a voice coding system, the following three basic requirements should be determined.

  1. The transmission speed of digital data.
  2. Delay due to the encoding process.
  3. Allowable losses, or the relative number of lost packets, before the voice quality, becomes below the threshold value.

Any method of converting voice to digital format degrades sound quality. Generally speaking, this deterioration of the signal is indistinguishable to the human ear. However, if some bits change due to transmission errors or are lost when discarding packets, this affects the quality of the recovered signal.

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Posted on: May 11, 2017Ana Nichols