Author :
Dvorský, Peter ; Londák, Juraj ; Lábaj, Ondrej ; Podhradský, Pavol
Author_Institution :
Fac. of Electr. Eng. & Inf. Technol., STU, Bratislava, Slovakia
Abstract :
The article is analyzing the real impact audio and video streams for teleconferencing services in NGN networks. The test scenario is based on examining the data flow between Asterisk PBX and SIP client. This examination is performed by analysis of the data stream for the Asterisk PBX, depending on the parameters for audio/video stream to the SIP client. These parameters are represented using different codecs for audio (G.711 a-law, G.711 u-law, G.722, GSM, Speex), and the video (H.263, H.263p, H.264, MP4V-ES) to required bandwidth. The second part is a subjective assessment of quality audio / video stream, depending on network parameters (delay, packet loss rates and bandwidth). The third part of looking at video conferencing is an analysis of CPU usage and RAM consumption, depending on the number of active SIP clients. Above all, however, affect the number of users in a conference room.
Keywords :
audio coding; audio streaming; next generation networks; signalling protocols; teleconferencing; video codecs; video streaming; Asterisk PBX; CPU usage analysis; G.711 a-law; G.711 u-law; G.722; GSM; H.263; H.263p; H.264; MP4V-ES; NGN networks; RAM consumption; Speex; active SIP clients; audio codecs; data flow; delay; network parameters; next generation network; packet loss rates; real-impact audio streaming; subjective assessment method; teleconferencing services; video codecs; video streaming; videoconferencing service; Bandwidth; Delay; GSM; Loss measurement; Streaming media; Video codecs; Asterisk; IMS (Internet Protocol Multimedia Subsystem) architecture; codecs; video conference;
