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- Create Date March 13, 2017
- Last Updated September 16, 2026
A Real Time Control Protocol RTCP 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 Control Protocol (RTCP) works alongside RTP to provide reporting and feedback about the quality and performance of real-time media streams. While RTP carries the voice packets themselves, RTCP can provide valuable information about packet loss, jitter, timing, round-trip delay, and the overall condition of a VoIP session. For engineers and technicians troubleshooting call-quality problems, that feedback can reveal issues that may not be obvious from examining RTP packets alone.
A dedicated Wireshark RTCP Profile makes this information easier to identify and interpret during packet analysis. By emphasizing RTCP Sender Reports, Receiver Reports, source identifiers, interarrival jitter, packet-loss statistics, timestamps, and delay information, the profile helps technicians quickly evaluate what the endpoints themselves are reporting about the media stream. Wireshark exposes these and other RTCP metrics directly in its protocol analysis.
For VoIP engineers and technicians, RTCP analysis provides another layer of evidence when investigating poor audio quality, delay, intermittent packet loss, or inconsistent media performance. Rather than relying solely on what Wireshark calculates from captured RTP traffic, the profile makes it easier to compare those observations with the quality information being exchanged by the VoIP endpoints themselves, creating a more complete picture of call performance.