Checking for Network Impairments of Incoming RTP Packets (moved)

Moved to ​https://docs.pjsip.org/en/latest/specific-guides/audio/checks/rx_quality.html

The pjsua's dq (dump quality of current call) command from pjsua's menu provides excellent tool for troubleshooting network impairments of incoming RTP packets.

>>> dq
 14:54:37.008        pjsua.c
  [CONFIRMED ] To: sip:user@localhost;tag=1857bde149264e2986c4aac1a26f5866
    Call time: 00h:00m:53s, 1st res in 1071 ms, conn in 1191ms
    #0 speex @16KHz, sendrecv, peer=192.168.0.66:4000
       RX pt=103, stat last update: 00h:00m:01.753s ago
          total 1.7Kpkt 72.2KB (127.2KB +IP hdr) @avg=10.6Kbps
          pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
                (msec)    min     avg     max     last
          loss period:   0.000   0.000   0.000   0.000
          jitter     :   0.000   5.506 227.000   9.812
       TX pt=103, ptime=20ms, stat last update: 00h:00m:07.871s ago
          total 0pkt 0B (0B +IP hdr) @avg 0bps
          pkt loss=0 (-1.$%), dup=0 (-1.$%), reorder=0 (-1.$%)
                (msec)    min     avg     max     last
          loss period:   0.000   0.000   0.000   0.000
          jitter     :   0.000   0.000   0.000   0.000
      RTT msec       :   0.274   0.616   1.233   1.233

The following information is provided by the above output.

Call Identification

The following line provides identification of the call:

  [CONFIRMED ] To: sip:user@localhost;tag=1857bde149264e2986c4aac1a26f5866

SIP Signaling Statistic

    Call time: 00h:00m:53s, 1st res in 1071 ms, conn in 1191ms

The line above tells us the duration of the call, the delay before the first non-100 response is received, and the delay before the call is confirmed.

Stream Identification

    #0 speex @16KHz, sendrecv, peer=192.168.0.66:4000

The line above identifies the first stream in the session (in the future, a session may have more than one streams, e.g. one audio stream and one video stream), the codec being used, the direction of the stream, and the address where RTP packets will be transmitted to.

RX Statistics

       RX pt=103, stat last update: 00h:00m:01.753s ago

The line above shows the expected RTP payload to be received from remote (103) and the time when the last RTCP SR/RR packet is sent.

          total 1.7Kpkt 72.2KB (127.2KB +IP hdr) @avg=10.6Kbps

The line above shows the total number of RTP packets and total number of RTP payload size that has been received since the media is established.

          pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)

The line above shows total number of packet loss, duplicate packets, and out-of-order packets respectively, along with their percentage against total received packets.

                (msec)    min     avg     max     last
          loss period:   0.000   0.000   0.000   0.000

The line above shows the minimum, average, maximum, and last duration of the loss packet, to see the maximum duration of a packet loss burst.

          jitter     :   0.000   5.506 227.000   9.812

The line above shows the minimum, average, maximum, and last jitter value of incoming RTP packets.

TX Statistics

The TX statistics are calculated from the RTCP SR/RR packet received from remote, so if remote doesn't support RTCP, the TX statistic values will be all zero.

The following line shows whether incoming RTCP SR/RR packet has been received from remote:

       TX pt=103, ptime=20ms, stat last update: 00h:00m:07.871s ago

The stat last update time above shows when the last time RTCP SR/RR was received from remote.

RTT Estimates

The RTT estimates is calculated from the RTCP RR received from remote. If no RTCP RR is received, the RTT would be displayed as zero.