Detecting Packet Loss in RTP Phone Calls Using Wireshark

  Рет қаралды 2,335

Plaintext Packets

Plaintext Packets

Күн бұрын

Пікірлер: 6
@unathimehlomakulu8540
@unathimehlomakulu8540 8 ай бұрын
Wonderful Vid….thank you for these wireshark nuggets. Can you do something about analyzing WebRTC calls in wireshark pretty please?
@daddyegaming
@daddyegaming 8 ай бұрын
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
@plaintextpackets 8 ай бұрын
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.
@karankashyap3969
@karankashyap3969 7 ай бұрын
Good 👍 wonderful vedeo
@xsTaoo
@xsTaoo 7 ай бұрын
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
@plaintextpackets 7 ай бұрын
RTP is normally used in large organizations like companies, schools, hospitals, etc.
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