In the case of a conference, create it on the Connect Client and dial in here, too, all participants in the conference must be known. Make a precise note of the participants of the call and the time. In the next step, set up the call via AnyDevice. The port range for the RTP stream is permanently set to 30000 to 33000 on the XPhone Connect Server.įirst start the recording of the network via Wireshark with the appropriate filters. If other ports are used for SIP communication on the XPhone server, these must be used in the Wireshark filter, e.g.: portrange 5060-5062 || portrange 30000-33000 Note the point under XPhone Call Controller Gateway: For example, the gateways or end devices of the telephone system can be stored here.īefore you start the recording via Wireshark, you must be aware that the call will be recorded and that you must inform all participants.Īn up-to-date Wireshark and the corresponding WinPcap must be installed on the server.įor the analysis of the voice quality, the following IP addresses must be known:įurthermore, the port for SIP communication must be taken into account so that the RTP packets can be assigned to the corresponding calls.Ĭheck the settings for SIP communication by opening the settings for the corresponding SIP trunk in the web interface of XPhone Connect Server under Telephony > Gateways > SIP. This information can be found in the SDP header. In the SIP Invite message and in the SIP OK of the remote terminal the information about the RTP stream is transmitted. The voice packets (RTP) do not have to take this route. The communication for call set-up via SIP takes place directly between the telephone system and the XCC SIP Trunk. the conversation partners cannot hear each other, there is one-sided communication or dropouts occur in the conversation. When using one of the two functions, problems occur in the voice quality, i.e. The analysis assumes that the function of AnyDevice / Softphone or the conference server of XPhone Connect is used.
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