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.* >>> >>> >> >> >> >> --------------------------------------------------------------------- >> To unsubscribe, e-mail: [email protected] >> For additional commands, e-mail: [email protected] >> >> >> > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [email protected] > For additional commands, e-mail: [email protected] > > -- Thank you Patrick Song Thinking globally, Networking locally CCVP, CCNP, M.Eng in Telecommunications Cell:1-647-868-2950
