Voip
Hardware/Coding
Many steps have to be taken to setup a VOIP communication. First, the ADC is needed to convert analog voice to digital signals (bits). Now the bits have to be compressed in a good format for transmission: there are a number of protocols that will be seen afterwards. Then, we have to insert our voice packets in data packets using a real-time protocol (typically RTP over UDP over IP). We need a signaling protocol to call users: ITU-T H323 does that. At RX we have to disassemble packets, extract data, then convert them to analog voice signals and send them to sound card or phone. (Valdes, Robert)
Testing
Network Operators must have an effective means for testing voice service quality if they are to attract and retain customers. This means testing beyond IP network performance measurements. End user speech quality measurements are needed to provide network operators with the level of visibility needed to effectively manage service quality. It is well known that the IP network performance parameters that impact speech quality are packet loss, delay, and jitter. But the type and degree of impact that these parameters have on speech quality are lesser known. This is because there are many other VoIP processes that impact speech, and these various processes, together with IP network performance conditions, influence each other in complex ways to affect speech quality. (Valdes, Robert) It is very easy for the complex interactions of these processes and conditions to get out of control and not only degrade speech quality, but do so in a way that is difficult to diagnose and analyze. Speech quality on traditional circuit-switched networks was never a major issue, so measurement techniques were limited. Electrical signal measurements, such as signal-to-noise ratio or total harmonic distortion, were generically applied to measuring speech quality as a strictly electrical signal. Today, with the deployment of next generation...
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