Detecting Packet Loss in RTP Phone Calls Using Wireshark

This video demonstrates how to detect packet loss in the RTP protocol using Wireshark. RTP (or Real-Time Transport Protocol) is a common network protocol which uses UDP for transport. It can be tricky to detect packet loss in UDP streams, so we cover how RTP sequencing works and how to use telephony analysis in Wireshark to measure stream quality.
Credit to wiki.wireshark.org for sample captures.

[@unathimehlomakulu8540]

Wonderful Vid….thank you for these wireshark nuggets.
Can you do something about analyzing WebRTC calls in wireshark pretty please?

[@karankashyap3969]

Good 👍 wonderful vedeo

[@daddyegaming]

Great video again and thank you! So what % of packet loss would you consider to say that there really is something bad going on and not just isolated because in an ideal enterprise, there'd still be some packet loss but to some extent we say there's really no issue.

[@plaintextpackets]

Within the enterprise (say internal voice calls) the threshold should be 0%, though that’s often difficult with branches. Proper QoS policies on all potential bottlenecks are a must to reach that target. That being said, packet loss as low as 1-5% can be noticeable to end users.

[@xsTaoo]

In what situations do people use RTP connections? I tried two mobile phones to talk in real time on chat software, but they don't seem to use RTP connections.

[@plaintextpackets]

RTP is normally used in large organizations like companies, schools, hospitals, etc.