VoIP voice quality

The main problem of IP telephony, is the loss of IP packet transmission in the network. In this publication, I would like to analyze the most common cases. Below is a diagram of a transmission path of voice under ideal conditions.

VOIP тракт без потерь

VOIP tract without loss

By and large, the problem of equipment on the transmission side to convert an analog voice signal, or a video in the numbered blocks in the picture, and in reality it is UDP packets with timestamps, or the so-called RTP stream. The task of the receiving path, assemble these building blocks in accordance with the time-stamped and recreate using the Digital-to-analog converter (DAC) in the copy of the signal transmitting side.

VOIP тракт с потерей пакетов

VOIP path with packet loss
In case of packet loss, lost part of the media information that causes breaks characteristic voice or monochrome squares in the image in the case of video content. Delivery speed RTP packets in IP networks are constantly changing, depending on the boot IP tract. This is called jitter. To compensate for the uneven speed of packets at the receiver side creates a temporary package store, or the so-called jitter buffer. His task is to collect incoming packets in the correct order according to the time stamp and give them the codec at regular intervals and the correct order.

jitter

Work jitter buffer
The size of the buffer receiving VOIP device calculates during operation or forcibly configured. On the one hand it can not be too high so as not to increase the transport delay. On the other hand, the small size of the buffer is packet loss with changes in the delay time in the IP network. Hence the one of the main contradictions between Internet providers and users of IP telephony. From the viewpoint of a provider, all packets delivered to the subscriber, i.e., there are no losses. And in terms of VOIP device, the time difference between the arrival of a package far exceeds the jitter buffer. Therefore, the actual losses are. From experience loss of more than 1% causes a certain discomfort. More than 2% of the conversation is difficult. For values greater than 4% conversation is almost impossible.

In order to show the possible causes of packet loss in real Internet network, consider the scheme a call from the softphone on a landline phone via the VOIP provider.

Схема звонка в IP телефонии

Driving a call to IP telephony

From subscriber to a local Internet service provider

The most common cause is the lack of capacity of the connecting line between the provider and the subscriber. The presence of parallel file downloads, even at extremely low speeds, can cause very serious problems, especially when the subscriber line capacity is less than 700Kb / s. The fact that the size of a standard package when downloading files from http or ftp is approximately equal to 1500 bytes. To pass this package through the channel capacity, say 128Kbit / s requires about 95 mc. Here, the situation can be compared to a sports car, tried to overtake a truck on the site with traffic in a row. That is, even if the priority queue for voice modems or routers on both sides, can seriously degrade the quality.

Another reason why the author has repeatedly noticed in the implementation of different kinds of call-manager-s, call centers, IP-PBX. Switch-It and routers such as “soap box” or ethernet cables of unknown quality. As well as the use of embedded routers ATA gateways to route traffic to other VOIP ATA gateways. The fact that the actual performance of different kinds of network devices is not measured in megabits, kilobits, gigabits per second, and in packets per second. By its very nature any ip phone generates a number of packets read, creates strain on network equipment, ten times more than downloading favorite series via bittorent. Accordingly, any networking equipment is no longer able to correct errors in the poor quality of SCS, or Switches and routers simply can not cope with the load.

The last reason most rare today is the lack of a computer processor. The work of two ACELP codec (transmit-receive), download 100% Pentium 200mhz processor. With today’s machines overload the CPU, is that it is possible to encode the video.

Problems on the Internet

There are usually two:

First, it is the greed of local providers not promptly increase the capacity of backbone providers connections. Moreover, in small ISPs is the practice of creating two or more traffic queues. One more priority, as a rule, fall games, web traffic and icmp packets. In less priority queue fall, usually, voip, torent and everything else. That is, from the point of view of an unsophisticated subscriber like they are fine, but in fact external channels are overloaded. By the way, for the same reason utilities ping and traceroute is not a good tool for assessing voice quality between servers on the Internet.

The second reason is quite trivial peer war. It usually serious clashes between backbone operators, terminators common interconnect regularly arise in connection with someone who has to pay for the traffic. If you are interested look for Ukrtelecom vs UA-IX, Golden Telecom vs RTCom, Cogent vs Level 3. Such problems usually last from several months to many years and cause serious problems for VOIP customers.

Internet telephony provider.

In practice, the provider, there are two schemes enable subscribers with full proxy and media proxy signaling. In the scheme with full proxy RTP RTP stream goes through the proxy server provider and it is possible the same set of problems as on the subscriber side. Typically, this overload RTP proxy servers and routers. Very often there are problems in small ISPs trying to start IP phone business. They are generally linked to the absence of certain priority queues for voice, or their incorrect setting. In the scheme with the signal proxies through server provider is signaling only that in no way affects the quality of voice and RTP stream goes directly from the subscriber to the subscriber. It should be noted that if one party is behind NAT, in which case the proxy signaling scheme ceases to operate in the case of mandatory use full proxy.

IP gateway in the telephone operator.

Typically, the licensed operators reserves channel capacity and stocks containers routers are so great that an overload practically impossible. There are following problems. In many countries, there are quite serious corruption, bureaucratic obstacles against getting VOIP traffic to the telephone network. Consequently, according to market laws, the price of white legitimate traffic and traffic generated by the black and gray schemes vary considerably. The provider of IP telephony when choosing a partner, not always has the ability or the desire to conduct an audit of the quality and in most cases chooses the offer with the lowest price. The operator terminating traffic black scheme does not care about the installation of quality equipment because of the risk of confiscation and does not have much choice online channel. Since, as a rule, the lifetime of such an operator is calculated for days or months at best, nobody really cares about the packet losses