<div dir="ltr"> They seem self-explanatory from the code:<br><div><br></div><div>        apt_log(APT_LOG_MARK,APT_PRIO_INFO,"Close RTP Receiver %s:%hu <- %s:%hu [r:%u l:%u j:%u p:%u d:%u i:%u]",<br>                        rtp_stream->rtp_l_sockaddr->hostname,<br>                        rtp_stream->rtp_l_sockaddr->port,<br>                        rtp_stream->rtp_r_sockaddr->hostname,<br>                        rtp_stream->rtp_r_sockaddr->port,<br>                        receiver->stat.received_packets,<br>                        receiver->stat.lost_packets,<br>                        receiver->rr_stat.jitter,<br>                        mpf_jitter_buffer_playout_delay_get(receiver->jb),<br>                        receiver->stat.discarded_packets,<br>                        receiver->stat.ignored_packets);<br></div><div><br></div><div>The baseline depends on the codec used too.</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, May 18, 2020 at 10:33 PM Unai Rodriguez <<a href="mailto:unairodcas@gmail.com">unairodcas@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">Dear List,<br><br>We're experiencing choppy/distorted audio streams at times and have been looking into understanding the definition and units of measure of these values reported on the main FS log (i.e. /var/log/freeswitch/freeswitch.log)<br><br>- r = received<br>- l = lost<br>- j = jitter<br>- p = playout_delay<br>- d = discarded_packets<br>- i = ignored_packets<br><br>The following log lines seemed to correspond with choppy/distorted audio:<br><br>========================================================<br>...<br>2020-05-15 10:18:05.202714 [INFO] mpf_rtp_stream.c:541 () Close RTP Receiver <a href="http://10.252.148.75:18800" target="_blank">10.252.148.75:18800</a> <- <a href="http://10.252.148.73:17592" target="_blank">10.252.148.73:17592</a> [r:403 l:0 j:651 p:0 d:87 i:0]<br>...<br>2020-05-15 10:25:53.622708 [INFO] mpf_rtp_stream.c:541 () Close RTP Receiver <a href="http://10.252.148.75:22660" target="_blank">10.252.148.75:22660</a> <- <a href="http://10.252.148.56:18572" target="_blank">10.252.148.56:18572</a> [r:402 l:0 j:143 p:0 d:84 i:0]<br>...<br>2020-05-15 10:43:42.642722 [INFO] mpf_rtp_stream.c:541 () Close RTP Receiver <a href="http://10.252.148.75:18940" target="_blank">10.252.148.75:18940</a> <- <a href="http://10.252.148.72:20800" target="_blank">10.252.148.72:20800</a> [r:356 l:0 j:838 p:0 d:79 i:0]<br>...<br>2020-05-15 11:00:52.102707 [INFO] mpf_rtp_stream.c:541 () Close RTP Receiver <a href="http://10.252.148.75:20952" target="_blank">10.252.148.75:20952</a> <- <a href="http://10.252.148.61:16540" target="_blank">10.252.148.61:16540</a> [r:582 l:0 j:806 p:0 d:66 i:0]<br>...<br>========================================================<br><br>At this point we believe that having a (d)iscarded value of more than 60 correlates with choppy calls and most of the time those come with (j)itter value of several hundreds.<br><br>Would it be possible for someone to point out where we could find the definition and unit of measure of those values (i.e.: r, l, j, p, d, i)? Is there any baseline that could be references to determine which calls are being choppy/distorted based on these numbers? We've been trying to understand FS source code and find documentation to no avail.<br><br>Thank you so much<br>Unai Rodriguez<br></div>
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