A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> a) RTP Multiplexing Techniques and Applications <s> IP telephony systems are expected to be deployed worldwide in the near future because of their potential for integrating the multimedia communication infrastructure over IP networks. Phone-to-phone connection over an IP network via IP telephony gateways (IP-GWs) is a key feature of the system. In an IP telephony system, a low-bit-rate voice codec is used to improve bandwidth efficiency. However, due to the packet transfer method over the IP network, it is necessary to add packet headers, including IP, UDP, and RTP headers, which increases the header overhead and thus decreases transfer efficiency. Moreover, because there will be large numbers of short voice packets flowing into the IP network, the load on the Internet will increase. We propose voice stream multiplexing between IP-GWs to solve these problems. In this scheme, multiple voice streams are connected between a pair of IPCWs, enabling multiplexed voice stream transfer. The voice stream multiplexing mechanism can reduce the header overhead as well as decrease the number of voice packets. The voice stream multiplexing we propose is to concatenate RTP packets destined for the same IP-GW at a multiplexing interval period into a single UDP packet. The advantage of this method is that no new additional header is required and the current well-defined H.323 and RTP standards can be applied with minimum changes. We implemented and tested the system. <s> BIB001 </s> A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> a) RTP Multiplexing Techniques and Applications <s> Voice communications such as telephony are delay sensitive. Existing voice-over-IP (VoIP) applications transmit voice data in packets of very small size to minimize packetization delay, causing very inefficient use of network bandwidth. This paper proposes a multiplexing scheme for improving the bandwidth efficiency of existing VoIP applications. By installing a multiplexer in an H.323 proxy, voice packets from multiple sources are combined into one IP packet for transmission. A demultiplexer at the receiver-end proxy restores the original voice packets before delivering them to the end-user applications. Results show that the multiplexing scheme can increase bandwidth efficiency by as much as 300%. The multiplexing scheme is fully compatible with existing H.323-compliant VoIP applications and can be readily deployed. <s> BIB002
Hoshi and his colleagues BIB001 , proposed a Voice Multiplexing scheme that could be used to reduce the number of RTP packets transported over the IP Network. Their scheme combines voice packets from varying streams into a single UDP packet for transmission. Although this technique improves transmission efficiency, it does not handle the idea of compressing the RTP headers BIB001 . This mechanism for saving bandwidth has been leveraged upon by Sze et al BIB002 , where the proposed a technique that explores the combination of packet multiplexing and header reduction through accumulating several RTP packets in a single UDP packet. The RTP packets are compressed for which context-mapping tables are created in the multiplexer and de-multiplexer. This is to ensure that original packets are rebuilt or restored at the destination end. Through experimentation / simulation, the proposed system showed an effective utilization of Wide Area Network trunk and an increase in the number of supported real-time calls severally, as compared to the conventional scheme BIB002 . A multiplexing scheme presented by  also expresses more light on the theory of RTP multiplexing using mixers and translators. While the mixer is tasked with collecting and adding-up of different voice streams, changing the data format and retransmitting to the receiver, the translator re-encodes multiple packets into one. In the long run, both the mixer and translator send only a combined or translated RTP flow. Thus, the TC RTP defines the combination of protocols for effective management of voice packets. ECRTP header compression scheme compresses the IP, UDP and RTP headers into one new header. Thus, final transmission of packets through respective tunnelling scheme is achieved using the PPP multiplexing.