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.

I believe that skip may 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.

But this is guesswork...


On 29 November 2012 20:35, Mick Stevens <mickstevens@yahoo.com> wrote:
Hi Michael &  Co,

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...?

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 rtp_audio_in_jb_packet_count & rtp_audio_in_largest_jb_size = jitter buffer? (apologies if everybody else already knows this!).

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! :) happy

    <rtp_audio_in_raw_bytes>1565066</rtp_audio_in_raw_bytes>
    <rtp_audio_in_media_bytes>1564856</rtp_audio_in_media_bytes>
    <rtp_audio_in_packet_count>9113</rtp_audio_in_packet_count>
    <rtp_audio_in_media_packet_count>9098</rtp_audio_in_media_packet_count>
    <rtp_audio_in_skip_packet_count>1609</rtp_audio_in_skip_packet_count>
    <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
    <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
    <rtp_audio_in_cng_packet_count>15</rtp_audio_in_cng_packet_count>
    <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
    <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>

Thank you in anticipation of enlightenment!

#:x lovestruckFreeSWITCH

From: Michael Collins <msc@freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Sent: Thursday, 29 November 2012, 19:14
Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?

Mick,

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 mod_xml_curl wiki page. For the record, here's a quick dump of the fields that I took from an XML CDR:

    <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes>
    <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes>
    <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count>
    <rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count>
    <rtp_audio_in_skip_packet_count>18</rtp_audio_in_skip_packet_count>
    <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
    <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
    <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count>
    <rtp_audio_in_flush_packet_count>11</rtp_audio_in_flush_packet_count>
    <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
    <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes>
    <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes>
    <rtp_audio_out_packet_count>67</rtp_audio_out_packet_count>
    <rtp_audio_out_media_packet_count>67</rtp_audio_out_media_packet_count>
    <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count>
    <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count>
    <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count>

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).

If anyone knows these please reply to this email and Mick and I will get them documented on the wiki (right Mick? ;)

rtp_audio_in_skip_packet_count
rtp_audio_out_skip_packet_count
rtp_audio_in_jb_packet_count
rtp_audio_in_flush_packet_count
rtp_audio_in_largest_jb_size

Thanks all!

-MC

On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <mickstevens@yahoo.com> wrote:
Hi Folks,

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?

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.

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.

Any help much appreciated!
 
Rgds, Mick

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_________________________________________________________________________
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_________________________________________________________________________
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http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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