Yes, Thank you much better Status : OK (25 ms) and Status : OK (20 ms)
On Mon, 2010-04-12 at 14:17 -0400, Philip Mullis 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]
