Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond....) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line about 2 Mb. We feel our quality decrease when in US are about 9:00 or 10:00 in the morning. We do not know if this is it correct or all the people using VoIp provider feel the same quality? Anyone knows any provider without this kind of problems? Witch provider do you use to get the best sounds quality? Any clue will be welcomed. Thanks for your time Obihuan.
On Tue, 2005-03-29 at 12:36 +0200, Ismael Gil wrote:> Hello all, > > We recently configure an asterisk server to use with an VoIP provider > to make calls to a PSTN. We use (voipjet, nufone, diamond....)If you find the same problem with multiple ITSP's, then it may not be them that is at fault.> We feel that we haven't got the quality that we hope. Sometimes our > calls gets mute, or we feel communication cuts on our phone calls. > We have got an QOS router (Draytek) reserving 1/2 of our wideband to > the SIP an IAX2 protocols, and an ADSL line about 2 Mb.Sounds like it should be quite adequate... how many simultaneous calls are you doing?> We feel our quality decrease when in US are about 9:00 or 10:00 in the morning.What time is that for your local time? Is there something that might be happening at/around that time for you? eg, here, around 3 - 6pm is quite busy as school kids get home and go on the internet, same for people getting home from work. In fact, my vague recollection is that things just get busier until around 11pm, before they really slow down. While this doesn't have any relation to *your* adsl connection, think about what this might be doing to your ISP's internet connection....> We do not know if this is it correct or all the people using VoIp > provider feel the same quality?Not that I would know, but I get the feeling that most people get extremely good quality calls over a decent internet connection.> Anyone knows any provider without this kind of problems? > Witch provider do you use to get the best sounds quality?I've not used any, so can't comment on this. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 0000 adam@websitemanagers.com.au Fax: +61 2 9345 4396 www.websitemanagers.com.au
> We recently configure an asterisk server to use with an VoIP provider > to make calls to a PSTN. We use (voipjet, nufone, diamond....) > > We feel that we haven't got the quality that we hope. Sometimes our > calls gets mute, or we feel communication cuts on our phone calls. > We have got an QOS router (Draytek) reserving 1/2 of our wideband to > the SIP an IAX2 protocols, and an ADSL line about 2 Mb.ADSL has slower upload speeds than download speeds (your 2Mbps is download). so you may have problems with your outgoing packets of sound. g.711 codec (the default codec for most voip providers because there is virtually no sound quality loss) uses about 84Kbps per channel or simultaneous connection. For example if you have an Upload speed of 128Kbps. and you try to have 2 phone conversations you would need 168kbps transfer speed. That is 40kbps more than your upload speed. This is a major problem with ADSL the upload and download speeds are not equal. Another potential problem is that your provider is over subscribed for the available bandwidth. What this means is that when allot of people are using their connection to your provider. The provider may not be able to handle all those users at once and packets get dropped or delayed. Dropping or delaying packets is very bad for VoIP especially if they do not do QoS or ToS routing which most providers do not. What is your upload speed? Some other possibilities are to use some compression codecs which will cause some sound quality loss like gsm or iLibc and g.729 to pack more calls in the limited bandwidth limitations. Another option is to use SDSL where the speeds of both the upload and download are the same.> We feel our quality decrease when in US are about 9:00 or 10:00 in the morning.This time is when businesses in the us are opening and starting to do business In the united states. Both for phones and Data.> We do not know if this is it correct or all the people using VoIp > provider feel the same quality?This may mostly be in relation to you Internet provider and how many hops you have to take to get to the VoIP provider and if they oversubscribe their bandwidth capacity. One provider may be good for one person with one person in a different ISP than an ISP you have. And you are even right next door to each other. This is as a result of how the internet is connected and may not nessessarly be geographic. For example you may be connecting to a server in your own city lets say Chicago but you are actually routed to San Francisco then back to Chicago. But it will not always take the same path the next time you may be routed through New York. This is a simplification of how it works. The closer you are to a Tier 1 provider(they own the major trunks interconnects) the less time it will take to get to your target.> Anyone knows any provider without this kind of problems?I have seen many Providers have both Good and bad connection links. It is best to have a provider that routes with QoS and/or ToS within their routers and have only one or two hops between your provider and a tear 1 provider.> Witch provider do you use to get the best sounds quality?It is not that simple. But you can begin by doing a traceroute to the many providers at different times of the day. This will see the route changes and time delays between hops to get to VoIP Providers gateways. Hope this helps in understanding the problems involved with choosing a provider. Thanks, Max
> It is not that simple. But you can begin by doing a traceroute to the > many providers at different times of the day. This will see the route > changes and time delays between hops to get to VoIP Providers gateways.The best tool I've found for monitoring connections, routes, congestion, is called PingPlotter. http://pingplotter.com/ It's a shareware visual traceroute. It continually graphs the traceroute style responses. There is a scrollable timeline to view how things change. You can get raw data out of it as well. It records changes in routes. Their web site also has some tutorials on how to use pingplotter to track down problems. Unfortunately it's windows only. It will run under vmware though. I have no affiliation with them. I've just found it very useful. --Rob
Ping runs as a low priority service so it is not realistic to measure response time using ping. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Johnathan Corgan Sent: Thursday, March 31, 2005 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoIP Provider problems Johnathan Corgan wrote:> First off, I have Sprint Broadband Direct internet service, a fixed > wireless setup with a 2-5 Mbps downlink and a terrible 128 kbpsuplink.> So I know I'm in for trouble anyway. > > The broadvoice edge router (63.251.209.126, their lax site) is another> 11 hops away. One hop before that, the packet loss rate has gone up to> 13%, so the Internet adds another 4% to my sucky ISP connection. Round> trip time to this point is 200ms, so-so but livable. > > Here's the kicker: > > Reported packet loss from broadvoice, one additional hop, is awhopping> 29%. So between the last "Internet" router (bbnet2.lax.pnap.net) and > broadvoice's edge router, there is an additional 16% loss.Just an update after about 12 hours of data--the data above was worst case. During off-peak hours in the middle of the night the packet loss at my ISP was effectively zero, and only 3% along the way to broadvoice, with a 75ms round-trip time. Broadvoice edge-router still reports 28% packet loss though, and an additional 30ms RTT increase for this last hop. So I even more strongly suspect (or just really hope) they are preferentially discarding non-RTP traffic in favor of voice traffic. I did discover that the multi-second outages are at my local ISP, not at Broadvoice--for some reason Sprint BBD can take up to 4 seconds to respond to a ping, so something is really wrong there--but is there a way to do this type of testing in a more rigorous and controlled fashion? -Johnathan _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users