A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> 1) Voice / Packet Header Compression <s> We describe a new method to multiplex a number of low bit rate audio streams into a single RTP stream between IP telephony gateways. Audio frames from different users are assembled into a multiplexed RTP payload thus reducing the overhead of RTP/UDP/IP headers. To maintain distinction between the multiplexed audio streams sharing a single RTP stream, a 2-byte mini-header is added to each frame of an audio stream. A unique channel identifier is associated between the peer gateway entities by means of signaling procedures. Major applications of this method will be in the areas of IP telephony gateways that interconnect PSTN/PBX systems and cellular access networks. Results have indicated that the overhead is reduced by 50% to 80% depending on the size of the audio frame. Other advantages of the proposed scheme include fewer UDP/IP connections between gateways, a reduced possibility of congestion at intermediate IP routers, and a reduction of processing delays in multiprotocol label switching (MPLS), differentiated service (diff-serv) and IP security (IPsec) environments. <s> BIB001 </s> A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> 1) Voice / Packet Header Compression <s> In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients <s> BIB002 </s> A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications <s> 1) Voice / Packet Header Compression <s> Header compression techniques can improve the speech quality of VoIP applications. However, its deployment over wireless mesh environments may not offer much improvement, since the wireless and multi-hop characteristics of such a network impose a high packet loss rate. In this work, we propose a new approach for header compression, using packet aggregation to cooperate with the header compression scheme. Simulation results show that the new cooperative approach can decrease significantly the packet loss, impacting positively on the speech quality. <s> BIB003
This refers to the phenomenal representation in which voice packets are compacted from the default 40 bytes size to a smaller 2 bytes size in best. Compression and decompression codecs are used to achieve this. Codecs convert the voice signals from analog to digital, and additionally compress the resultant digital voice using digital compression algorithms. The compressed information is transformed to frames (packet payload) of varying sizes depending on the type of codec being used and its efficiency BIB001 . Header compression for bandwidth optimization is motivated by some basic actualities. First, VoIP datagram is typically compressed at the application layer, implying that no other compression is required to condense payload size. Secondly, a huge portion of the data packet size is occupied by the headers, thirdly, significant redundancies usually exist in IP headers which could be best managed via compression means with recourse to efficient algorithms and(or) protocols BIB002 . These protocols as would be seen do help reduce greatly the size of data, and there are numerous compression codecs with varying compression rates as could be seen in table 1 below. The key backings for header compression is the reduction of the amount of control information in packet headers and subsequently decreasing its portion in the overall packet size, and increasing bandwidth availability for data (voice/payload) transportation. Redundant details of each flow are captured and stored in a data structure on both the source (compressor) and destination (de-compressor) BIB003 . Quite clear to grasp is that header compression exploits the VoIP packet field characteristics of either being unchanged or increasing at persistent ratio all through a call period. Compression/transmission protocols (CRTP, ROCCO, RoHC, etc.) are what help to effectively ensure source to destination movement of voice packets. With the help of such protocols as would be reviewed further in this work, this technique does meaningfully improve bandwidth utilization. Figure 3 shows a pictorial representation of the compression technique. 