Simulation of VoIP over UDP with Bandwidth on Demand Analysis Sandeep Kamra, Kunal Gupta Amity University, Noida [email protected]Abstract This paper analyses resource provisioning for enterprise Voice-over-IP (VoIP) networks. Simulation and analytical methods are used to enhance the provisioning process, which ultimately aims to provide the Bandwidth and its availability. This paper defines simple guidelines for network dimensioning in a multimedia environment in terms of end-to-end delay for the voice traffic, and in terms of throughput and packet loss for TCP data traffic. A realistic environment is modelled and simulated using ns-2. The model consists of a prioritized network in which intermediate routers perform priority scheduling to provide differentiation of Internet services. 1. Introduction The increase in both popularity and capacity of the Internet has led to the increasing need to provide real-time voice and video services to the network. While the potential benefits of these services are enormous, the process of adapting the connectionless data-oriented design of IP networks to real-time traffic is rather slow. Recently, we have seen two major trends in the area of communications. First, IEEE 802.11 WLANs have been widely deployed in the world. Second, due to the growth of Internet bandwidth, real-time audio and video applications have become more mature and popular. The combined effect has made VoIP (voice over IP) over WLANs possible. Although VoIP involves the transmission of digitized voice in packets, the telephone itself may be analog or digital. The voice may be digitized and encoded either before or concurrently with packetization. 2. G.711 G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in 1972. Its formal name is Pulse code modulation (PCM) of voice frequencies. It is required standard in many technologies, for example in H.320 and H.323 specifications. It can also be used for fax communication over IP networks (as defined in T.38 specification). G.711, also known as Pulse Code Modulation (PCM), is a very commonly used waveform codec. G.711 uses a sampling rate of 8,000 samples per second, with the tolerance on that rate 50 parts per million (ppm). Non-uniform (logarithmic) quantization with 8 bits is used to represent each sample, resulting in a 64 kbit/s bit rate. G.711 fax passthrough does not distinguish between a G.711 voice call and a fax call, treating both the same way and not doing anything in particular to address the problems of packet loss, jitter and delay. The fax message is carried in its entirety in-band over the voice call. This is a technique that is tried by many businesses to send faxes over the Internet since all they need is the same G.711 standard support that they already have for VoIP calls. G.711 passthrough works over LANs or networks that do not suffer packet losses or excessive delays. It does not work reliably over the open Internet. Some customers‟ mistake G.711 passthrough for all VoIP techniques, which often has them, staying away from Internet fax due to lower reliability. G.711 is a voice standard, but G.711 passthrough is not a fax standard. 3. G.729 With the low rate of 8 kbps, G.729 is the lowest bit rate ITU-T standard with toll quality, offering opportunities for significant increases in bandwidth utilization in existing telephony and wireless applications. G.729 operates on 10-ms frames, allowing moderate transmission delays, so applications such Sandeep Kamra et al, International Journal of Computer Science & Communication Networks,Vol 2(2), 284-287 284 ISSN:2249-5789
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Simulation of VoIP over UDP with Bandwidth on Demand Analysis
