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- Create Date March 13, 2017
- Last Updated October 6, 2026
A Real Time Protocol (RTP) profile for VoIP analysis in Wireshark
This profile is for VoIP analysis in Wireshark. You need a number of profiles to be successful with VoIP packet analysis.
Real-time Transport Protocol (RTP) carries the actual audio media for most traditional VoIP calls, making it one of the most important protocols for voice engineers and technicians to understand. When users report choppy audio, one-way audio, gaps in speech, or inconsistent call quality, examining the RTP streams can provide the packet-level evidence needed to determine what is really happening.
A dedicated Wireshark RTP Profile helps bring the most useful voice-media information to the surface quickly. By focusing the packet view on RTP sequence numbers, timestamps, payload information, packet timing, jitter, loss, and stream behavior, the profile makes it easier to identify missing packets, out-of-sequence traffic, timing irregularities, and other conditions that can affect perceived voice quality.
For VoIP engineers and technicians, the benefit is a faster and more repeatable troubleshooting workflow. Instead of rebuilding columns, filters, and display settings for every investigation, the profile provides a purpose-built environment for examining RTP traffic, correlating packet behavior with call-quality complaints, and documenting the evidence needed to distinguish media problems from signaling or underlying network issues.