![]() A progress bar indicates the position in the stream and is synchronized amongst all RTP streams that are played. Pressing the "Play" button plays the RTP stream from within Wireshark. Note that all RTP packets that are dropped because of the jitter buffer are reported ("Drop by Jitter Buff"), as well as the packets that are out of sequence (Out of Seq). You can now see all RTP streams available for the calls that you selected: The jitter buffer emulated by Wireshark is a fixed size jitter buffer and can efficiently be used to reproduce what clients can effectively hear during the VoIP call. To play the RTP audio stream of one or multiple calls from the VoIP List, select them from the list and then press the "Player" button:Ĭhoose an initial value for the jitter buffer and then press the "Decode button". To check your Wireshark follow this procedure: The official builds contain all of the plugins maintained by the Wireshark developers, but custom/distribution builds might not include some of those codecs. The codecs supported by Wireshark depend on the version of Wireshark you're using. Prior to version 3.2.0, it only supported saving audio using the G.711 codec from 3.2.0 it supports saving audio using any codec with 8000 Hz sampling. Wireshark allows you to save decoded audio in. Wireshark allows you to play any codec supported by an installed plugin. When clicking a packet in the Graph, the selected frame will be selected in the Main Wireshark window.
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