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

Timestamp:

Dec 5, 2006 8:11:46 PM (17 years ago)

Author:

bennylp

Comment:

--

audio-check-packet-loss

@@ +1 @@
+= Checking for Network Impairments of Incoming RTP Packets =
+
+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.
+