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]
