Enhancing VoIP BW Utilization over ITTP Protocol

Abstract

The revolution of Voice over Internet Protocol (VoIP) technology has propagated everywhere and replaced the conventional telecommunication technology (e.g. landline). Nevertheless, several enhancements need to be done on VoIP technology to improve its performance. One of the main issues is to improve the VoIP network bandwidth (BW) utilization. VoIP packet payload compression is one of the key approaches to do that. This paper proposes a new method to compress VoIP packet payload. The suggested method works over internet telephony transport protocol (ITTP) and named Delta-ITTP method. The core idea of the Delta-ITTP method is to find and transmit the delta between the successive VoIP packet payloads, which is typically smaller than the original VoIP packet payload. The suggested Delta-ITTP method implements VoIP packet payload compression at the sender side and decompression at the receiver side. During the compression process, the Delta-ITTP method needs to keep some values to restore the original VoIP packet payload at the receiver side. For this, the Delta-ITTP method utilizes some of the IP protocol fields and no additional header is needed. The Delta-ITTP method has been deployed and compared with the traditional ITTP protocol without compression. The result showed that up to 19% BW saving was achieved in the tested cases leading to the desired enhancement in the VoIP network BW utilization.

An innovative approach for enhancing capacity utilization in point-to-point voice over internet protocol calls

“…As previously stated, successive packets increase the Ti and SN elements by a constant value. Assume that the Ti and SN elements for five consecutive packets are (3,10), (6,20), (9,30), (12,40), and (15, 50), respectively. Clearly, depending on the numerical relation between them, the values of the Ti element can be determined from the corresponding values of the SN element.…”

“…These protocols are 12 bytes of real-time transfer protocol (RTP), 8 bytes of user diagram protocol (UDP), and 20 bytes of IP protocol (40 bytes of RTP/UDP/IP). Therefore, the wasted network capacity, based on the size of the codec audio segment and RTP/UDP/IP protocols, can reach up to 80% [4]- [6]. Generally, VoIP calls can be divided into point-to-point (P-P), point-to-multipoint, and multipoint-tomultipoint.…”

Tel-MPLS: a new method for maximizing the utilization of IP telephony over MPLS networks

“…At this point, it is possible to say that each call is aware of the socket on the other end. During the media transmission phase, a separate protocol, such as RTP, is used to send voice data [8], [33]. RTP, along with user datagram protocol (UDP) and IP, transfers voice data to its intended destination, utilizing the call factors from the initial phase, including the source socket.…”

“…With an internet connection, it is possible to make phone calls without the need for traditional phone service or copper lines. To make calls, one needs solely a high-speed internet connection and an IP telephony service provider [6]- [8]. Due to the aforementioned characteristics, MPLS technology has helped promote IP telephony [9]- [11].…”

Enhancing Bandwidth Utilization of IP Telephony Over IPv6 Networks

“…The digital speech samples are typically between 10-byte to 30-byte, as shown in Tab. 1 [3,4]. In certain cases, the VoIP packet payload consists of more than one voice frame.…”