Voice over Internet Protocol (VoIP) has been an interesting topic of research in the last decade. The engrossing increase in the use of VoIP services is resulting in the enormous growth of broadband network. The main objective of this paper is the selection of an appropriate voice compression and decompression (CODEC) schemes depending on the Quality of Service (QoS) of VoIP in different networks. Wired, Wireless Local Area Network (WLAN), Worldwide Interoperability for Microwave Access (WiMAX) and Universal Mobile Telecommunication System (UMTS) networks were implemented in OPNET Modeler. The quality is compared using different QoS parameters like end-to-end delay, MOS, throughput and jitter. The VoIP codecs used in the measurements of QoS are: GSM-FR, G.711, G.723.1 and G.729A. Simulations showed that G.711 and GSM-FR are the best schemes that provide high quality of voice in Wireless Local Area Network (WLAN) communications. In WiMAX, G.729A gives the best quality of VoIP while in UMTS, GSM-FR gives overall best results with respect to all the parameters. Wired model gives the best result irrespective of the codec being used. G.723.1 can be used in WiMAX and UMTS along with the wired network depending on conditions. The results analyzed and the performance evaluated will give network operators an opportunity to select the codec for better services of VoIP for customer satisfaction.

Voice over Internet Protocol (VoIP) is an advanced area for researchers. Many different methods are used to send voice over IP networks. With the development of modern telecommunications equipments and softwares telecommunication's malpractices are growing rapidly. Hence there is always a need for monitoring communications and guarantee both security and proper usage. This underlined research work stresses on the analysis of IP traffic and proposes an algorithm for detection mechanisms to control and limit VoIP's grey traffic. The algorithm emphasizes primarily on Session Initiation Protocol (SIP) but it can be modified and used for all VoIP protocols like H.323 and InterAsterisk eXchange protocol (IAX2). The suggested method is based on analyzing the pcap files. These files are used to filter VoIP traffic from network's total IP traffic by reading the header of each packet. The algorithm then extracts different parameters for generating call logs. VoIP packets of the same call are correlated to produce a Call Detail Record (CDR).The produced CDR contains the IP addresses of source and destination that make the calls. For identification of grey traffic these IP addresses are used. If the source IP address in the CDR is of a legal registered operator the user/call is declared as legal, otherwise the user/call is illegal.