Detecting Packet Loss in RTP Phone Calls Using Wireshark

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  • เผยแพร่เมื่อ 23 ธ.ค. 2024

ความคิดเห็น • 6

  • @unathimehlomakulu8540
    @unathimehlomakulu8540 10 หลายเดือนก่อน +1

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

  • @daddyegaming
    @daddyegaming 11 หลายเดือนก่อน

    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  11 หลายเดือนก่อน +1

      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
    @xsTaoo 9 หลายเดือนก่อน

    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  9 หลายเดือนก่อน

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

  • @karankashyap3969
    @karankashyap3969 10 หลายเดือนก่อน

    Good 👍 wonderful vedeo