Subscribers who use long-distance communication services expect from it the usual “telephone” quality, so the quality of voice communication is a key success factor in this sphere.
While choosing the VoIP provider, one should take into account and ask the candidates about the QoS, the company provider should be competent in the development of technologies and protocols that improve the quality of voice in IP networks. The following sections of this document describe the functions and methods that determine the quality of voice transmission and play an important role in developing quality services.
What is Responsible for the Voice Quality At The Gateway?
In 1999, Mier Communications Company had compared the voice quality of VoIP systems based on the gateways from different manufacturers. As the figures showed, the Cisco solution showed the highest quality, exceeding the minimum requirements for the quality of traditional long-distance and international telephony services.
The high-quality voice at the gateway is achieved through advanced processing techniques and ensuring a minimum delay. These methods reduce bandwidth requirements, reduce costs and ensure the timely transmission of voice packets.
The quality of voice communication is largely dependent on voice processing at the gateway. The Cisco gateway supports a variety of encoding and decoding tools (CODEC) and has the following voice processing capabilities:
It supports standard CODEC G.729 (CS-ACELP), G.729a, G.723.1 and G.711 (PCM) tools that allow high-quality voice transmission over low-bandwidth channels (up to 8 kbps). To measure the quality of the connection, an average score or Mean Opinion Score (MOS) is used, which is calculated from the feedback of subscribers, who evaluate the quality on a five-point scale from 1 (unsatisfactory quality) to 5 (distortions are imperceptible). G.729 CODEC has an average MOS score of 4.4 and G.711 CODEC at 4.2. Both means correspond to the level of quality of long-distance telephone communication, which should be at least 4.2 points, and G.729 even exceeds it.
The gateway performs the function of suppressing the echo, which allows you to get rid of the back noise. For this, the E.165 Echo cancellation standard is used.
The gateway generates a so-called “comfort noise”. As a result, despite the suppression of pauses (and, as a result, bandwidth savings), we avoid the unpleasant “dead silence” in the handset, in those periods when voice packets are not transmitted through the line.
Some additional VoIP parameters can be adjusted to achieve a more acceptable quality of communication. Among them, the gain and attenuation of the signal, the depth of echo cancellation, the level of recognition of voice activity, the signal-to-noise ratio and the degree of background noise suppression.
Minimizing Delays Is The First Must Do Technique To Provide The Best Quality Of Voice Communications
Delays are the most important factor affecting the quality of voice communications. For applications related to real-time voice transmission, ITU recommends delays in 150 ms or less. Working in conjunction with QoS, the gateway is able to minimize latency throughout the channel due to several factors:
• Large delays are prevented by the performance of voice encoding/decoding processes.
• The intelligent adaptive vibration buffer smooths the fluctuations in the length of delays in the transmission of packets.
• The gateway can be configured for IP Precedence, which ensures the priority of voice packets in backbone queues.
• Multiclass Multilink PPP fragmentation and interleaving (RFC 1990) breaks up volumetric data packets that are insensitive to the time factor, into smaller fragments in order to integrate voice Packets of real-time. This technology is especially useful for low-speed channels. It does not allow large data packets to block real-time voice packets.
• Resource Reservation Protocol (RSVP) allows the gateway to request network bandwidth and reserve it.
Minimizing Bandwidth Requirements
As already noted, G.729 CODEC supports high-quality voice communications even in low-speed channels (8 kbit/s). Cisco VoIP gateways use the following tools to minimize bandwidth requirements:
To transmit faxes, only 9.6K of bandwidth per second is needed.
Incoming facsimile messages are demodulated in advance, which allows reducing the bandwidth requirements arising from the use of 64-kilobit pulse-code modulation.
The gateway has the functions of recognizing voice activity and suppressing pauses. As a result, the bandwidth is used only at those times when one of the subscribers speaks. During pauses (which account for about half of the entire talk time), the bandwidth is available for other traffic. The maximum time required to recognize a pause is 200 ms.
The Compressed Real-Time Transfer Protocol (CRTP) compresses a conventional 40-byte header to 2 to 4 bytes. Reducing the packet header length by 90 percent gives a very big benefit, especially in the case of real-time voice packets, which differ in their small size. This technology is recommended for low-speed channels.