Trial testing monitoring

Discussion in 'Web Call Server 5' started by mbedial, May 11, 2017.

  1. mbedial

    mbedial New Member

    Hi,
    we have a trial with a client. We have implemented several sip accounts using the phone_min android app, and our client (an important mobile operator) is testing it.
    The client is reporting some issues most of them related to bad quality of the audio, disconnections and some times no audio in one of the direction of the call. My question is , what's the best way to monitor these issues? Do the logs help us to debug quality topics?

    Thanks in advacne
  2. Max

    Max Administrator Staff Member

    The best way is analyzing of RTP / SRTP media traffic of the WebRTC-SIP call.

    [​IMG]
    WebRTC / SRTP - here
    SIP / RTP - here
    You can see packet lost, jitter, wrong sequence errors in the dump.
    Code:
    tcpdump udp -i any -s 4096 -w log.pcap

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