Changes between Initial Version and Version 1 of audio-check-packet-loss


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Timestamp:
Dec 5, 2006 8:11:46 PM (14 years ago)
Author:
bennylp
Comment:

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  • audio-check-packet-loss

    v1 v1  
     1= Checking for Network Impairments of Incoming RTP Packets = 
     2 
     3The pjsua's '''{{{dq}}}''' (dump quality of current call) command from pjsua's menu provides excellent tool for troubleshooting network impairments of incoming RTP packets. 
     4 
     5{{{ 
     6>>> dq 
     7 14:54:37.008        pjsua.c 
     8  [CONFIRMED ] To: sip:user@localhost;tag=1857bde149264e2986c4aac1a26f5866 
     9    Call time: 00h:00m:53s, 1st res in 1071 ms, conn in 1191ms 
     10    #0 speex @16KHz, sendrecv, peer=192.168.0.66:4000 
     11       RX pt=103, stat last update: 00h:00m:01.753s ago 
     12          total 1.7Kpkt 72.2KB (127.2KB +IP hdr) @avg=10.6Kbps 
     13          pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) 
     14                (msec)    min     avg     max     last 
     15          loss period:   0.000   0.000   0.000   0.000 
     16          jitter     :   0.000   5.506 227.000   9.812 
     17       TX pt=103, ptime=20ms, stat last update: 00h:00m:07.871s ago 
     18          total 0pkt 0B (0B +IP hdr) @avg 0bps 
     19          pkt loss=0 (-1.$%), dup=0 (-1.$%), reorder=0 (-1.$%) 
     20                (msec)    min     avg     max     last 
     21          loss period:   0.000   0.000   0.000   0.000 
     22          jitter     :   0.000   0.000   0.000   0.000 
     23      RTT msec       :   0.274   0.616   1.233   1.233 
     24}}} 
     25 
     26The following information is provided by the above output. 
     27 
     28== Call Identification == 
     29 
     30The following line provides identification of the call: 
     31{{{ 
     32  [CONFIRMED ] To: sip:user@localhost;tag=1857bde149264e2986c4aac1a26f5866 
     33}}} 
     34 
     35== SIP Signaling Statistic == 
     36 
     37{{{ 
     38    Call time: 00h:00m:53s, 1st res in 1071 ms, conn in 1191ms 
     39}}} 
     40 
     41The 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. 
     42 
     43== Stream Identification == 
     44{{{ 
     45    #0 speex @16KHz, sendrecv, peer=192.168.0.66:4000 
     46}}} 
     47The 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. 
     48 
     49== RX Statistics == 
     50{{{ 
     51       RX pt=103, stat last update: 00h:00m:01.753s ago 
     52}}} 
     53The 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'''. 
     54 
     55{{{ 
     56          total 1.7Kpkt 72.2KB (127.2KB +IP hdr) @avg=10.6Kbps 
     57}}} 
     58The 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.  
     59 
     60{{{ 
     61          pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) 
     62}}} 
     63The line above shows total number of packet loss, duplicate packets, and out-of-order packets respectively, along with their percentage against total received packets. 
     64 
     65{{{ 
     66                (msec)    min     avg     max     last 
     67          loss period:   0.000   0.000   0.000   0.000 
     68}}} 
     69The line above shows the minimum, average, maximum, and last duration of the loss packet, to see the maximum duration of a packet loss burst. 
     70 
     71{{{ 
     72          jitter     :   0.000   5.506 227.000   9.812 
     73}}} 
     74The line above shows the minimum, average, maximum, and last jitter value of incoming RTP packets. 
     75 
     76 
     77== TX Statistics == 
     78 
     79The 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. 
     80 
     81The following line shows whether incoming RTCP SR/RR packet has been received from remote: 
     82 
     83{{{ 
     84       TX pt=103, ptime=20ms, stat last update: 00h:00m:07.871s ago 
     85}}} 
     86 
     87The stat last update time above shows when the last time RTCP SR/RR was received from remote. 
     88 
     89 
     90== RTT Estimates == 
     91 
     92The RTT estimates is calculated from the RTCP RR received from remote. If no RTCP RR is received, the RTT would be displayed as zero. 
     93