Hi Darry
I assume the dsl service you have is in the residential area based on
what you said "the quality gets worse in the evening". is this
correct? it is because the traffic flow pattern for business
environment is high during the day and is less at night.
one of the reasons you have worse quality is that the uplink from
dslam to provider network is fully overbooked in your area.
also, the TOS value you set in the IP packets is ignored by the dslam
because dslam is a layer 2 device and it looks at MAC address only
by the way, Internet is best effort service for most service providers
and no guaranteed speed, delay and jitter (NO SLA)
thank you
On Mon, Apr 12, 2010 at 2:17 PM, Philip Mullis <[email protected]>
wrote:
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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--
Thank you
Patrick Song
Thinking globally, Networking locally
CCVP, CCNP, M.Eng in Telecommunications
Cell:1-647-868-2950
