Mike Diehl
2018-Dec-21 20:13 UTC
[asterisk-users] Question about packet counts in voipmonitor
Hi all, I'm not sure this is the place to ask, but here goes... I'm using voipmonitor to gather call statistics such as packet counts, average jitter, etc. Eventually, I want to use those stats to detect and alert on poor call quality. However, I'm finding that the packet counts for each leg of a given call can vary quite a bit. For example, I have a call that was connected for 84 seconds. At 50 frames/sec, I expect to see about 4200 frames. However, on one side I see 4187 (which is good) and on the other side, I only see 2577 frames sent. Am I doing something wrong? Or is this approach simply doomed? Any thoughts would be welcome. -- Mike Diehl -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20181221/79c0a3ea/attachment.html>
Dovid Bender
2018-Dec-25 09:46 UTC
[asterisk-users] Question about packet counts in voipmonitor
Mike, Are you using port mirroring or is VoipMonitor running on the same box? If the latter I would run tcpdump and compare what VM says it has to what you see in your wireshark dump. If you are sniffing via port mirroring your switch maybe dropping packets (we had that when we tried to mirror 700 mbit of traffic on a Juniper EX4200). On Fri, Dec 21, 2018 at 8:13 PM Mike Diehl <mdiehlenator at gmail.com> wrote:> Hi all, > > > > I'm not sure this is the place to ask, but here goes... > > > > I'm using voipmonitor to gather call statistics such as packet counts, > average jitter, etc. Eventually, I want to use those stats to detect and > alert on poor call quality. > > > > However, I'm finding that the packet counts for each leg of a given call > can vary quite a bit. > > > > For example, I have a call that was connected for 84 seconds. At 50 > frames/sec, I expect to see about 4200 frames. However, on one side I see > 4187 (which is good) and on the other side, I only see 2577 frames sent. > > > > Am I doing something wrong? Or is this approach simply doomed? > > > > Any thoughts would be welcome. > > > > -- > > Mike Diehl > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20181225/850976a4/attachment.html>