New RTP Packet Payload Shrinking Method to Enhance Bandwidth Exploitation Over RTP Protocol

Abstract

One of the pillars to run businesses is the telecommunications. Most of the institutions are migrating, if not already migrated, to Voice over Internet Protocol (VoIP) technology. However, VoIP still need some improvements, in terms of networks bandwidth exploitation and VoIP call quality, to meet the businesses expectations. Networks bandwidth exploitation, which is our concern in this paper, has been enhanced using different approaches and methods. This paper suggests a new method to enhance networks bandwidth exploitation Packet's payload shrinking (compression) approach. The suggested method works with the RTP protocol and called RTP Payload Shrinking (RPS) method. As the name implies, the RPS method will reduce the size of the RTP packet payload, through shrinking it based on specific algorithm, which enhances the networks bandwidth exploitation. The RPS method utilizes the RTP fields to store the values that are needed to apply the shrinking algorithm at the sender and receiver sides. The effectiveness of the proposed RPS method has been examined in comparison to conventional RTP protocol without shrinking. The deployment result showed that the saved bandwidth ratio has reached up to nearly 17% in the tested scenarios. Therefore, enhancing the network bandwidth exploitation.

Using the Packet Header Redundant Fields to Improve VoIP Bandwidth Utilization

“…Then, the original value of the Timestamp can be derived from the Sequence-number field of the RTP header at the PM-CHr module. The Timestamp and Sequence-number fields are augmented by constant values for a series of packets [7,8,17]. For instance, assume the Sequence-number a series of packets is 2, 4, 6, 8 and, 10.…”

“…Assuming 10 bytes voice segment, then the packet header will consume 80% of the available bandwidth. Thus, a huge waste of IP network bandwidth [1,7,8]. Various range of multimedia applicant uses RTP to transfer their data.…”

“…Some other VoIP developers have created the 4 bytes IAX protocol to carry the voice data instead of the 12 bytes RTP protocol. This also has saved the network bandwidth to some extent [7,12,13]. For instance, assuming 10 bytes voice segment, then the packet header, when using 32 bytes IAX/UDP/IP, will consume 64% of the available bandwidth.…”

Contracting VoIP Packet Payload Down to Zero

“…To transfer the voice media, VoIP technology uses so-called media transfer protocols [1,10]. Real-time Transport Protocol (RTP) is the dominant and most used media, both voice and video, transfer protocol [11,12]. The RTP protocol works in collaboration with the UDP protocol to be able to convey the voice data over the IP network.…”

“…The size of the RTP protocol is 12 bytes, the UDP protocol is 8 bytes, and the IP protocol is 20 bytes. Therefore, the VoIP packet header size comprises of 40 bytes RTP/UDP/IP [12,13]. On the other hand, the VoIP codecs produce a very small voice frame (VoIP packet payload) between 10 and 30 bytes.…”