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How do I analyse voice quality problems?



The analysis assumes that the function of AnyDevice / Softphone or the conference server of XPhone Connect is used.

When using one of the two functions, problems occur in the voice quality, i.e. the conversation partners cannot hear each other, there is one-sided communication or dropouts occur in the conversation.


The communication for call set-up via SIP takes place directly between the telephone system and the XCC SIP Trunk. The voice packets (RTP) do not have to take this route. In the SIP Invite message and in the SIP OK of the remote terminal the information about the RTP stream is transmitted. This information can be found in the SDP header. For example, the gateways or end devices of the telephone system can be stored here.



Before you start the recording via Wireshark, you must be aware that the call will be recorded and that you must inform all participants.

An up-to-date Wireshark and the corresponding WinPcap must be installed on the server.

For the analysis of the voice quality, the following IP addresses must be known:

  • Telephone system
  • Gateways
  • IP terminals
  • XCC SIP Trunk

Furthermore, the port for SIP communication must be taken into account so that the RTP packets can be assigned to the corresponding calls.

XPhone Connect Server

Check 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. Note the point under XPhone Call Controller Gateway:


Wireshark on the XPhone Connect Server

Open Wireshark and use the following filters before starting the trace:

port 5060 || portrange 30000-33000


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

The port range for the RTP stream is permanently set to 30000 to 33000 on the XPhone Connect Server.

Recording for analysis via Wireshark

First start the recording of the network via Wireshark with the appropriate filters.

In the next step, set up the call via AnyDevice. Make a precise note of the participants of the call and the time.

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. You should also note the exact time here.

Continue the conversation until the expected misbehaviour occurs, also note the exact time of the misbehaviour.

Afterwards, end the conversation and stop the recording in Wireshark. Before you start the analysis, the created trace should be saved.

Analysis of the Wireshark Trace

If the Wireshark trace is available, you can start the analysis. In Wireshark you have the option to listen to the recorded conversation. Here you can listen to each stream that was sent to the XPhone Connect Server or which streams the XPhone Connect Server sent to the individual call partners. You can retrieve this information using the source and destination address in the IP protocol.

Telephony -> VOIP calls

On the one hand, you can check the VOIP calls via the item Telephony à VoIP Calls. There you can also see the course of the call setup and the RTP stream. Check the parameters for correctness and whether the correct subscribers/terminal devices/gateways were used. Furthermore, the option to listen to the individual streams is available here.

Telephony -> RTP Stream

Via the item Telephony -> RTP Stream all registered RTP streams are displayed. The information on packet loss, jitter and delta is shown in tabular form. For example, a packet loss with the codec G.711 of 1.0% can lead to a limited speech quality. Even in this case, short dropped calls occur.

Checking for QoS

The created Wiresharktrace also reveals the Quality of Service information of the packets. You should compare this with the settings in your network and check whether the appropriate values are set.

For example, the DSCP value "EF" is expected here, but this has been changed by active components in the network. This change affects the processing of the packets in the network, so that they can also be lost.


Please also refer to our documentation on the subject of QoS:

Checking the ptime

If it has already been checked whether QoS has been set correctly everywhere, the ptime should be checked. To do this, look in the message body of the SIP/SDP packets in Wireshark and search for the media attribute "ptime". XPhone has configured a ptime of 20. This cannot be changed. Deviations on the part of the telephone system can be the cause of poor voice quality.

Example: a ptime of 30 is configured in the OpenScape 4000.


Solution: The ptime must be configured to 20 by the telephone system.

For the OS4K you will find the setting "Frame Size" under "Explorers > Voice Gateway > Codec Parameters".

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