jb will indeed be jitterbuffer, but not sure what it'll mean. Discarded by jb, or left in jb at end of call maybe? Perhaps grepping the source for the tag name'll shed some light.<div><br></div><div>I believe that skip <i>may</i> be packets ignored because FS isn't ready to receive them yet. Eg receiving RTP on a call where nothing's consuming it, or on a bridged call that hasn't been answered yet.</div>
<div><br></div><div>But this is guesswork...</div><div class="gmail_extra"><br><br><div class="gmail_quote">On 29 November 2012 20:35, 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><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 class="im"><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><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 class="im"><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>
Skype: <a rel="nofollow">mick_stevens</a><br><a rel="nofollow">www.facebook.com/mickstevens</a><br><a rel="nofollow">www.twitter.com/mickstevens</a></font><br></div><div><br></div> </div><div style="font-family:arial,helvetica,sans-serif;font-size:12pt">
<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><div><div class="h5"> <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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