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This is the way I do it. Note this is for SIP calls!!!
In Wireshark select the Telephony TAB, the RTP then RTP streams. Look at the stream that has packet loss and note the two ports used for Source and Destination. In the Display filter type the filter: sdp.media.port==32136 (where the port number is one of the ports associated in step above (stream that has packet loss. I used 32136 as example). Now double click on the listed packet to open it up. Look for the "Generated Call-ID" and apply as a filter (not just Call-ID) This will then show the complete call with SIP/SDP/RTP
