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]
