Teh heh. Thanks Phil, I'll look at your site cheers, darryl
On Mon, 2010-04-12 at 14:41 -0400, Philip Mullis wrote: > (phil rolls eyes and points to himself) > > > Darryl Moore wrote: > > Thanks Philip. > > > > So...... > > > > do you know any VOIP providers that support VPN? I haven't explicitly > > looked for this service, but I certainly don't remember seeing it > > either. > > > > cheers, > > darryl > > > > On Mon, 2010-04-12 at 14:34 -0400, Philip Mullis wrote: > > > >> Also Darryl, one thing to point out this is relative latency to your > >> provider (if set to yes, is checked every 60 seconds) > >> > >> You can also use tools mtr to check for things like packet loss and > >> average latancys over all the hops... this will give you a better > >> snapshot.. note too though sometimes busy routers drop icmp so if you > >> see packet loss on 1 hop don't be too alarmed unless it carries over > >> multiple points... > >> > >> Alternately if your concerned that poor quality may be a result of > >> network management practices, try wrapping your connection in a vpn to > >> your provider if supported, networks like bells here in canada allow for > >> much more consistent throughput on certain types of traffic than others... > >> > >> Regards, > >> > >> Phil > >> > >> > >> > >> > >> Darryl Moore wrote: > >> > >>> 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] --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
