In vs Out are the 'listening' and 'talking' directions of the channel. They're not the same audio, and as such there's no reason they would match up. Couple that with sending/receiving before the other side, and network jitter/packet loss so that some packets arrive at different times or not at all.<div>
<br></div><div>As you point out using codecs can also affect them, as different codecs/ptimes will use different amounts of data/packets for the same time segment. Especially if the codec uses a VBR.</div><div><br></div>
<div>
Even using the 'echo' app and matching codec settings they will differ because of timing issues.</div><div class="gmail_extra"><br><br><div class="gmail_quote">On 5 December 2012 10:54, Mick Stevens <span dir="ltr"><<a href="mailto:mickstevens@yahoo.com" target="_blank">mickstevens@yahoo.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:12pt;font-family:arial,helvetica,sans-serif"><div><div><div style="font-size:12pt;font-family:arial,helvetica,sans-serif">
<div><span>Hey Team,</span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span><br></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span>Sorry to go on about this, but really keen to
understand these RTP fields in the XML CDR's better...</span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span><br></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span>1) At the risk of sounding too desperate... Please, please, please can someone explain to me what the <rtp_audio_in_skip_packet_count> & <rtp_audio_out_skip_packet_count> fields indicates?</span></div>
<div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span><br></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span>2) Also, for example in the extract below, why are the rtp_audio_in & rtp_audio_out bytes / packet counts not the same? Is this a red herring based on some codec interworking issue or similar, an indication of packet loss or expected/normal behaviour?</span></div>
<div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span><br></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span><div style="background-color:transparent"><span style="font-weight:bold"> <rtp_audio_in_raw_bytes>436848</rtp_audio_in_raw_bytes></span></div><div style="background-color:transparent"><span style="font-weight:bold"> <rtp_audio_in_media_bytes>436536</rtp_audio_in_media_bytes></span></div>
<div style="background-color:transparent"><span style="font-weight:bold"> <rtp_audio_in_packet_count>2562</rtp_audio_in_packet_count></span></div><div style="background-color:transparent"><span style="font-weight:bold"> <rtp_audio_in_media_packet_count>2538</rtp_audio_in_media_packet_count></span></div>
<div style="background-color:transparent"> <rtp_audio_in_skip_packet_count>1904</rtp_audio_in_skip_packet_count></div><div class="im"><div style="background-color:transparent"> <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count></div>
<div style="background-color:transparent"> <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count></div></div><div style="background-color:transparent"> <rtp_audio_in_cng_packet_count>24</rtp_audio_in_cng_packet_count></div>
<div class="im"><div style="background-color:transparent"> <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count></div><div style="background-color:transparent"> <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size></div>
</div><div style="background-color:transparent"><span style="font-weight:bold"> <rtp_audio_out_raw_bytes>624570</rtp_audio_out_raw_bytes></span></div><div style="background-color:transparent"><span style="font-weight:bold"> <rtp_audio_out_media_bytes>624360</rtp_audio_out_media_bytes></span></div>
<div style="background-color:transparent"><span style="font-weight:bold"> <rtp_audio_out_packet_count>3645</rtp_audio_out_packet_count></span></div><div style="background-color:transparent"><span style="font-weight:bold"> <rtp_audio_out_media_packet_count>3630</rtp_audio_out_media_packet_count></span></div>
<div class="im"><div style="background-color:transparent"> <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count></div><div style="background-color:transparent"> <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count></div>
</div><div style="background-color:transparent"> <rtp_audio_out_cng_packet_count>15</rtp_audio_out_cng_packet_count></div><div><br></div></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span>Please help! </span><img alt=":) happy" style="font-size:12pt"></div><div class="im"><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span> </span></div>
<div></div><div> </div><div><font face="arial, helvetica, sans-serif">Rgds, Mick</font></div><div><font face="arial, helvetica, sans-serif">Tel/SMS. <a href="tel:%2B44%280%297967%20594432" value="+447967594432" target="_blank">+44(0)7967 594432</a><span style="font-weight:bold"><br>
</span>Fax. <a href="tel:%2B44%280%297053%20452429" value="+447053452429" target="_blank">+44(0)7053 452429</a></font><font><span style="font-size:18px;color:rgb(0,51,102);font-weight:bold"></span><br></font></div><div><font face="arial, helvetica, sans-serif">Email/IM. </font><font><a rel="nofollow"><font face="arial, helvetica, sans-serif">mickstevens@yahoo.com</font></a><br>
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<div style="font-family:'times new roman','new york',times,serif;font-size:12pt"> <div dir="ltr"> <font face="Arial"> <hr size="1"> <b><span style="font-weight:bold">From:</span></b> Mick Stevens <<a href="mailto:mickstevens@yahoo.com" target="_blank">mickstevens@yahoo.com</a>><div class="im">
<br> <b><span style="font-weight:bold">To:</span></b> FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>> <br> </div><b><span style="font-weight:bold">Sent:</span></b> Thursday, 29 November 2012, 20:35<div>
<div class="h5"><br> <b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?<br> </div></div></font> </div><div><div class="h5"> <br><div><div>
<div style="font-size:12pt;font-family:arial,helvetica,sans-serif"><div><span>Hi Michael & Co,</span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span><br></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span>Many thanks for the prompt & positive response! Yes, more than happy to collate feedback from the global team & update the wiki accordingly (I'd like to be able to document what the field values between the > < indicate as well as just explain the field names if possible...) +anything else I can do to to contribute to the project...?</span></div>
<div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span><br></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span>Yes,
now the wiki is back up I've managed to work out that cng_packet = comfort noise generation! Also from the wiki, possibly that the jb in </span><span style="font-family:'times new roman','new york',times,serif;font-size:12pt">rtp_audio_in_jb_packet_count & </span><span style="font-family:'times new roman','new york',times,serif;font-size:12pt">rtp_audio_in_largest_jb_size = jitter buffer? (apologies if everybody else already knows this!).</span></div>
<div style="font-style:normal;font-size:12pt;background-color:transparent;font-family:'times new roman','new york',times,serif"><span style="font-family:'times new roman','new york',times,serif;font-size:12pt"><br>
</span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif"><span style="font-family:'times new roman','new york',times,serif;font-size:12pt">To provide the background to my original enquiry, I'm trying to identify if the rtp_audio_in_skip_packet_count in the following XML CDR extract is indicative of packet loss, or perhaps more accurately noticeable audio loss (as the packets haven't been lost, just ignored/"skipped"?: If anybody knows please speak up! </span><img alt=":) happy" style="font-family:arial,helvetica,sans-serif;font-size:12pt"></div>
<div style="font-style:normal;font-size:12pt;background-color:transparent;font-family:'times new roman','new york',times,serif"><span><br></span></div><div style="font-size:16px;font-family:'times new roman','new york',times,serif">
<span style="font-family:'times new roman','new york',times,serif;font-size:12pt"><div style="font-style:normal;background-color:transparent"> <rtp_audio_in_raw_bytes>1565066</rtp_audio_in_raw_bytes></div>
<div style="font-style:normal;background-color:transparent"> <rtp_audio_in_media_bytes>1564856</rtp_audio_in_media_bytes></div><div style="font-style:normal;background-color:transparent"> <rtp_audio_in_packet_count>9113</rtp_audio_in_packet_count></div>
<div style="font-style:normal;background-color:transparent">
<rtp_audio_in_media_packet_count>9098</rtp_audio_in_media_packet_count></div><div><span style="background-color:transparent"> </span><span style="font-weight:bold;text-decoration:underline;font-style:italic;background-color:rgb(230,230,230);color:rgb(255,0,0)"><rtp_audio_in_skip_packet_count>1609</rtp_audio_in_skip_packet_count></span></div>
<div style="font-style:normal;background-color:transparent"> <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count></div><div style="font-style:normal;background-color:transparent"> <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count></div>
<div style="font-style:normal;background-color:transparent"> <rtp_audio_in_cng_packet_count>15</rtp_audio_in_cng_packet_count></div><div style="font-style:normal;background-color:transparent">
<rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count></div><div style="font-style:normal;background-color:transparent"> <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size></div>
<div style="font-style:normal;background-color:transparent"><br></div></span></div><div style="font-style:normal;font-size:12pt;background-color:transparent;font-family:'times new roman','new york',times,serif">
Thank you in anticipation of enlightenment!</div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif"><br></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span>#</span><img alt=":x lovestruck" style="font-size:12pt">FreeSWITCH</div><div></div><div> </div><div><font face="arial, helvetica, sans-serif">Rgds, Mick</font></div><div><font face="arial, helvetica, sans-serif">Tel/SMS. <a href="tel:%2B44%280%297967%20594432" value="+447967594432" target="_blank">+44(0)7967 594432</a><span style="font-weight:bold"><br>
</span>Fax. <a href="tel:%2B44%280%297053%20452429" value="+447053452429" target="_blank">+44(0)7053 452429</a></font><font><span style="font-size:18px;color:rgb(0,51,102);font-weight:bold"></span><br></font></div><div><font face="arial, helvetica, sans-serif">Email/IM. </font><font><a rel="nofollow"><font face="arial, helvetica, sans-serif">mickstevens@yahoo.com</font></a><br>
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<div style="font-family:'times new roman','new york',times,serif;font-size:12pt"> <div dir="ltr"> <font face="Arial"> <hr size="1"> <b><span style="font-weight:bold">From:</span></b> Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>><br>
<b><span style="font-weight:bold">To:</span></b> FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>> <br> <b><span style="font-weight:bold">Sent:</span></b> Thursday, 29 November 2012, 19:14<br>
<b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?<br> </font> </div> <br><div>Mick,<br><br>This data is definitely not on the wiki - or anywhere else that I can see. I think we can crowdsource this to get the info collected and then add it to the <a rel="nofollow" href="http://wiki.freeswitch.org/wiki/Mod_xml_cdr" target="_blank">mod_xml_curl wiki
page</a>. For the
record, here's a quick dump of
the fields that I took from an XML CDR:<br>
<br> <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes><br> <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes><br> <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count><br>
<rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count><br> <rtp_audio_in_skip_packet_count>18</rtp_audio_in_skip_packet_count><br> <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count><br>
<rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count><br> <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count><br> <rtp_audio_in_flush_packet_count>11</rtp_audio_in_flush_packet_count><br>
<rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size><br> <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes><br> <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes><br>
<rtp_audio_out_packet_count>67</rtp_audio_out_packet_count><br> <rtp_audio_out_media_packet_count>67</rtp_audio_out_media_packet_count><br> <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count><br>
<rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count><br> <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count><br><br>I think raw_bytes, media_bytes, packet_count, and media_packet_count are self-explanatory. I think cng_packet_count is probably self-explanatory too. My question on dtmf_packet_count would be whether it's only for RFC2833 packets (I suspect yes, but would like confirmation). <br>
<br>If anyone knows these please reply to this email and Mick and I will get them documented on the wiki (right Mick? ;)<br><br>rtp_audio_in_skip_packet_count<br>rtp_audio_out_skip_packet_count<br>rtp_audio_in_jb_packet_count<br>
rtp_audio_in_flush_packet_count<br>rtp_audio_in_largest_jb_size<br><br>Thanks all!<br><br>-MC<br><br><div>On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <span dir="ltr"><<a rel="nofollow" href="mailto:mickstevens@yahoo.com" target="_blank">mickstevens@yahoo.com</a>></span> wrote:<br>
<blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:12pt;font-family:arial,helvetica,sans-serif"><div><span>Hi Folks,</span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span><br></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span>I'm trying to use FS XML CDR's to diagnose historic audio problems. I think I can work out some of the rtp_audio_in / out fields (raw bytes & media bytes being nearly equal looks like a good sign) but am wondering about the skip, cng & flush fields for example?</span></div>
<div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span><br></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span>I have tried Googling this & can find evidence of other people having asked this question but not of the answer. I have also checked my FS 106 & Cookbook book's without success. The wiki appears to be down at the moment so my apologies if the answer lies there.</span></div>
<div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span><br></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span>I know how to do this in real time using wireshark etc but am interested in being able to do some analysis on historic problems reported by customers that aren't willing/able to replicate the problem in order for a protocol trace to be
captured.</span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif"><span><br></span></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:arial,helvetica,sans-serif">
<span>Any help much appreciated!</span></div><div></div><div> </div><div><font face="arial, helvetica, sans-serif">Rgds, Mick</font></div><div><font face="arial, helvetica, sans-serif">Tel/SMS. <a rel="nofollow">+44(0)7967 594432</a><span style="font-weight:bold"><br>
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