Darryl, might say Unmonitored because your missing qualify=yes in that
providers sip profile.
Phil
Darryl Moore wrote:
Thanks Reza.
That is interesting.
One of the VOIP providers yields:
Status : OK (37 ms)
The other one says:
Status : Unmonitored
I wonder why one says unmonitored.
As I said, it doesn't get noisy until the evening. I expect my upstream
data is bottle necked at the DSLAM, I use the QoS bits in the IP packet,
but I'd be very surprised if Ma Bell actually looks at these. Especially
at the DSLAM.
I built a little Perl script to monitor the line which you can see at
http://moores.ca/qosplot.pl. This generally tells my if the latency is
due to the VOIP provider or the DSL. What I can't reliably figure out
from this, is if the latency is on the ATM network or the ISP network,
but I would certainly say it does not appear to be on the VOIP.
Note the data is collected by a different machine on my network from the
asterisk server. The asterisk server always has a higher priority, so
when my network gets busy (as it did this morning) VOIP generally does
not suffer, but my monitor will. I need to move it to run on the
asterisk box itself to be more accurate.
cheers,
darryl
On Mon, 2010-04-12 at 13:56 -0400, Reza - Asterisk Consultant wrote:
*Darryl:*
Please do a "sip show peer _your_trunk_provider" and let us know what your
latency is. 200ms is nothing in terms of a delay/lag between two human
voice conversations. I have people connecting to our platform from
overseas at 350ms+ latency **without** any jitter buffer enabled and quality
of connection is excellent. Their 350ms+ though seems to be huge (in
Toronto standards) - the connection we have between here and overseas office
is strong and stable (without congestion).
I am happy to give you a test account and DID on our server to help you
identify whether its a problem at your side, or whether the problem
magically goes away when you are connected with us.
" *Jitter is generally caused by congestion in the IP network. The
congestion can occur either at the router interfaces or in a provider or
carrier network if the circuit has not been provisioned correctly. *" --
so the trick here is to determine where the congestion is taking place.
Do at speed and VoIP quality check on the following:
1) http://myvoipspeed.visualware.com/servers/yul.html
2) http://myspeed.visualware.com/servers/yul.html
and share with us your stats.
>From the summary section, we would like to know your:
a) Connection Jiitter in ms
b) Packet Loss
c) MOS
We would also like to know your upload/download speed (of course). Along
with this, please copy and paste (except your password & userid) - your
entry you made in the sip.conf file in order to connect to your provider.
Kindly also share with us your DSL or Cable internet provider name.
The answers to the above will help determine where the fault is. Either
way - these issues are 100% solvable, assuming your carrier or ISP is
cooperative **if** we determine the problem is at their end.
*Best,
Reza.*
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