*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.* -- Toronto based VoIP / Asterisk Trainer, I.T. Consultant and Hosted PBX Solutions Provider. +1-647-476-2067. http://www.linkedin.com/in/seminar On Mon, Apr 12, 2010 at 12:35 PM, Darryl Moore <[email protected]> wrote: > So I am still trying to get my VOIP over DSL as reliable as possible. > > I've enabled variable jitter buffers of up to 200mS on all inbound audio > streams, figuring that a bit of delay is preferable to dropped packets. > Generally this works very well, so even when my line quality is > relatively poor, I have no problem hearing the party at the other end. > > I've spent a bit of time talking to my VOIP provider to see if they also > do or can implement variable buffers to help compensate for poor line > quality in the other direction. > > Their first response was that they do not need buffers as they are > connected directly to their service provider who in trn is connected > directly to the PSTN. I pointed out that the buffers would be useful, > not between them and their provider, but between them and ME. They said > they'd pass my concern up to their technical department. That is where > this has sat for a couple weeks now. > > Does anyone know much about VOIP providers? Do they tend to do much in > the way of jitter buffers on the audio streams they receive from their > customers? Are there any that are better than other? Perhaps letting > their customers customize the buffer parameters themselves? > > I have times (particularly in the evening) when the jitter can be quite > bad so a buffer of 200mS would be very useful. Other times there is > little jitter so a buffer of 50mS would be sufficient. It would be great > to know if my VOIP provider; first uses buffers at all; second, allows > variable buffers as I have with my asterisk setup. > > cheers, > darryl > > > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [email protected] > For additional commands, e-mail: [email protected] > >
