Re: [Asterisk-Users] VoIP Provider problems
No, I'm not ignorant of how this works. You'll notice I put it appears bad when I posted my results. Yes, it's not a perfect way to show problems -- but taken with a grain of salt it's not half bad. Especially when sampled over a longer period of time, and if the original poster can correlate the PingPlotter results to the quality of his calls. Now if he shows 30% loss during good and bad calls, that's another story. I posted my results to help the original poster. If he's trying to troubleshoot an apparent bad connection with Sprint, he needs all the help he can get. If they can proove the connection works even the littlest bit, they'll say it's fine and blame Broadvoice. If everyone gets similar levels of loss at those points, one could conclude its a side effect of the routers having better things to do. But if he's the only one showing them, then it would be a starting point to conclude something is wrong with his connection or something along Sprint's backbone. I'm not the original poster either, but for those following this thread keep in mind that a fair number of isp's use an upper-layer device to throttle data flows to some predeteremined rate. For example, I know some cable broadband companies that throttle their users to 128k up and some other value down. Don't have a clue whether their throttling box drops packets, delays them, or what; however, considering they would want to handle both udp and tcp, I'd have to bet some amount they drop udp packets to throttle udp data flows. On the other hand, I know of several dsl broadband companies that don't pay any attention to their uplink congestion, letting their uplink routers drop packets, etc. Since they can't afford to chase uplink utilizations by augmenting bandwidth, dropped packets happen frequently. Nature of the beast for some. ___ 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
Re: [Asterisk-Users] VoIP Provider problems
On Friday 01 April 2005 04:28, Joseph Gutowski wrote: Ok, since I guess no one else wanted to bite -- I will. I installed PingPlotter, switched to UDP just to be the same as you, and ran it against sip.broadvoice.com. Absolutley no problems, no packet loss at all. Ran it with all of the published proxy addresses, again no problems. I then used the 63.251.209.126 that you posted, and it was awful (at least it appears awful). I have reliable 20% packet loss at each of two Verio hops (nothing lost at the far end). Don't take this the wrong way, but you are showing a bit of ignorance about how TCP/IP works. The apparent packet loss you are seeing may be just fine tuning of the routers in question. The routers may be set up not to send ICMP host/network unreachables back to the originating system if they are required to send more than one in a configured time period. Routers have better things to do than continually tell you that a host is unreachable. B ___ 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
Re: [Asterisk-Users] VoIP Provider problems
Bob Goddard wrote: The apparent packet loss you are seeing may be just fine tuning of the routers in question. This is the conclusion I came to as well; however, with the way PingPlotter works the router is not sending ICMP unreachables but rather ICMP TTL expired responses. In any case, the routers in question may either be: 1) ...intentionally discarding the received UDP ping packets (these are not ICMP pings, but rather UDP packets with TTL down to zero when they get to the router), because the router has better things to do. 2) ...throttling the ICMP TTL expired responses to a certain rate per period of time, as you suggest. This would appear as packet loss. 3) ...actually congested, with the received UDP pings (and other types of packets) getting discarded on the input side at the rate shown in the data. I wish there was a way to measure 3) without being affected by 1) and 2). I agree then, that PingPlotter is not a highly accurate way to measure path quality. Still, though, looking over the data for a couple days now it is easy to see cyclical patterns that go from 1% to 30% (PingPlotter measured) loss, and an easily seen correlation with the voice quality of my outbound Broadvoice calls. Interestingly enough, switching from a Firefly soft phone on my workstation, using IAX2/ulaw, to an analog phone-TDM400 FXS port right at the Asterisk server has made a big difference. So some of the perceived crappiness was in the soft phone-Asterisk path and was probably being exacerbated by the network loss on the net or at Broadvoice's router. -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
Re: [Asterisk-Users] VoIP Provider problems
The apparent packet loss you are seeing may be just fine tuning of the routers in question. This is the conclusion I came to as well; however, with the way PingPlotter works the router is not sending ICMP unreachables but rather ICMP TTL expired responses. In any case, the routers in question may either be: 1) ...intentionally discarding the received UDP ping packets (these are not ICMP pings, but rather UDP packets with TTL down to zero when they get to the router), because the router has better things to do. 2) ...throttling the ICMP TTL expired responses to a certain rate per period of time, as you suggest. This would appear as packet loss. 3) ...actually congested, with the received UDP pings (and other types of packets) getting discarded on the input side at the rate shown in the data. I wish there was a way to measure 3) without being affected by 1) and 2). The deceptive part of doing the above is that once you see congestion (lack of an icmp response), you still have absolutely no idea what device was at fault. In other words, as the ttl value is increased and additional icmps are sent, you might see what you believe is congestion, but you still don't have any clue as to whether hop #2, #5, or #10 actually was involved with that congestion. ___ 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
Re: [Asterisk-Users] VoIP Provider problems
Rich Adamson wrote: In other words, as the ttl value is increased and additional icmps are sent, you might see what you believe is congestion, but you still don't have any clue as to whether hop #2, #5, or #10 actually was involved with that congestion. Sure. But there is a way around this. The traceroute-style statistics gathering technique that PingPlotter uses tries all the hops at the same time and plots the return rate for each one. So a 10 hop path has 10 packets go out, with individual packet's TTL set to expire at each hop. Done over and over again and averaged over many probes, you get a very clear picture. Packet loss at one node affects all the probes to that node and further ones, resulting in an increasing loss rate as you go down the path. For example: Hop Loss 1 0% 2 1% 3 1% 4 5% 5 5% 6 6% 7 15% 8 15% 9 16% 10 16% It's easy to see there is a big problem between hops 6 and 7 and a smaller problem between hops 3 and 4. With the broadvoice router I was seeing (at first) a jump from 0% to 9% at my local ISP, then small increments over the next 10 hops until it was at about 14%, then a big jump to 29% at the last hop. The data has varied cyclically between as high as the above and as low as 1% all the way across. Right this very moment, it is 2% within my ISP, still 2% all the way to PNAP, then a jump to 14% at the broadvoice ingress router at PNAP. Again, temper the above with the fact that the packet loss may be intentional, and these statistics not representative of real RTP traffic, as per my previous message. But I can predict with high accuracy what the caller on the other end of my broadvoice call will say about my voice quality based on the last number I see for the broadvoice ingress router. -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
Re: [Asterisk-Users] VoIP Provider problems
No, I'm not ignorant of how this works. You'll notice I put it appears bad when I posted my results. Yes, it's not a perfect way to show problems -- but taken with a grain of salt it's not half bad. Especially when sampled over a longer period of time, and if the original poster can correlate the PingPlotter results to the quality of his calls. Now if he shows 30% loss during good and bad calls, that's another story. I posted my results to help the original poster. If he's trying to troubleshoot an apparent bad connection with Sprint, he needs all the help he can get. If they can proove the connection works even the littlest bit, they'll say it's fine and blame Broadvoice. If everyone gets similar levels of loss at those points, one could conclude its a side effect of the routers having better things to do. But if he's the only one showing them, then it would be a starting point to conclude something is wrong with his connection or something along Sprint's backbone. ___ 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
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 kbps uplink. 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 a whopping 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
RE: [Asterisk-Users] VoIP Provider problems
Ping runs as a low priority service so it is not realistic to measure response time using ping. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 kbps uplink. 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 a whopping 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 ___ 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
Re: [Asterisk-Users] VoIP Provider problems
On Thursday 31 March 2005 15:59, Kellner, Peter wrote: Ping runs as a low priority service so it is not realistic to measure response time using ping. Try tracepath. It's not using port 7 and can be used by normal users. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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
Re: [Asterisk-Users] VoIP Provider problems
Ok, since I guess no one else wanted to bite -- I will. I installed PingPlotter, switched to UDP just to be the same as you, and ran it against sip.broadvoice.com. Absolutley no problems, no packet loss at all. Ran it with all of the published proxy addresses, again no problems. I then used the 63.251.209.126 that you posted, and it was awful (at least it appears awful). I have reliable 20% packet loss at each of two Verio hops (nothing lost at the far end). I did traceroutes on all of the Broadvoice proxies, and I didn't get pushed through PNAP. I wonder why your packets seem to reliably following that path when it's so bad. I mean the whole point of routing through PNAP is to increase quality, no? And from my understanding they're supposed to have a magic fuzzy logic to dynamically reroute around problems. Your results suggest a more widespread problem than one customer can't have nice VoIP calls -- you'd think Sprint wouldn't be routing through PNAP. Am I going to slap myself on the forehead in two minutes when I realize I missed something obvious and I'm completely off base here due to lack of sleep? I think you need to have a nice chat with Sprint, because it looks like your connection is pretty icky for anything, nevermind VoIP. I hope it's cheap. And I also hope your VoIP connection is wired if you're getting 9-10% loss on the wireless before you even leave the LAN. If you're starting off with a loss, it's just going to make the natural losses on the net have an even worse effect. ___ 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
Re: [Asterisk-Users] VoIP Provider problems
Joseph Gutowski wrote: I installed PingPlotter, switched to UDP just to be the same as you, and ran it against sip.broadvoice.com. Absolutley no problems, no packet loss at all. Well, that's good to hear. I then used the 63.251.209.126 that you posted, and it was awful (at least it appears awful). I have reliable 20% packet loss at each of two Verio hops (nothing lost at the far end). Okay, happy to see independent confirmation of this. I did traceroutes on all of the Broadvoice proxies, and I didn't get pushed through PNAP. I wonder why your packets seem to reliably following that path when it's so bad. I mean the whole point of routing through PNAP is to increase quality, no? And from my understanding they're supposed to have a magic fuzzy logic to dynamically reroute around problems. Your results suggest a more widespread problem than one customer can't have nice VoIP calls -- you'd think Sprint wouldn't be routing through PNAP. Well, you're right, Sprint is going through sprintlink.net - PNAP - BV, no route changes during the day since I started. Not a lot I can do about that, unfortunately. And I also hope your VoIP connection is wired if you're getting 9-10% loss on the wireless before you even leave the LAN. If you're starting off with a loss, it's just going to make the natural losses on the net have an even worse effect. It appears I happened to pick the most congested time to measure, and got 8-9% packet loss on my Sprint uplink. That's the wireless, as in a fixed wireless MMDS rooftop dish link to a mountain top about 15 miles away. It turns out that off peak there is zero loss over this link and typically it is only about 2-3% loss. So it's not as bad as it first seemed. On the premises it is all wired and first router is always zero packet loss. As I write this the trailing 10 minutes of data shows an aggregate 9% loss to BV with 3% of that on the Sprint BBD uplink side. This is much better than my first tests, and my SIP calls through broadvoice show the difference too. Anyway, I haven't tried the other broadvoice proxies yet, I'm really hoping at least one doesn't have PNAP on its path. (At least I can be thankful I haven't run into any of the weird NAT or authentication issues that have been discussed--worked great first time.) At the time I got this wireless link (which with a 4Mbps downlink, is pretty sweet for typical traffic patterns), there was no DSL in my area. Now SBC has service, but I've yet to really look into it. (As an aside, got my TDM400 card today, installed, and have the FXS port working with an analog phone. Woohoo. FXO next!) -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
RE: [Asterisk-Users] VoIP Provider problems
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 ___ 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
RE: [Asterisk-Users] VoIP Provider problems
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 ___ 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
Re: [Asterisk-Users] VoIP Provider problems
Give me a try! www.shelltel.com And don't use G711 for your calls. invest in the G729 codec. you'll find your calls will start working better. I'm a G729 shop. Thanks Michael D. Schelin 626-814-2454 Max W Blackmer Jr wrote: 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 ___ 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 ___ 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
Re: [Asterisk-Users] VoIP Provider problems
Robert Terzi wrote: 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. Thanks for the excellent link. I've had Asterisk on a home network and Broadvoice for a couple weeks now. IAX2 calls between Firefly soft-phone on my desk and other soft phones directly on the net have worked fairly well, but reported voice quality when going out over broadvoice to the PSTN has really stunk, making it only marginally useful. So I've downloaded this utility and am now tracing out sip.broadvoice.com, using UDP (as my ISP filters icmp.) Actually, the trace doesn't get past broadvoice's edge router, so I replaced the final IP address with that of the edge router itself so I could see the data instead of destination unreachable. Anyway, with only 20 minutes I've data I'm seeing some rather disappointing results. First off, I have Sprint Broadband Direct internet service, a fixed wireless setup with a 2-5 Mbps downlink and a terrible 128 kbps uplink. So I know I'm in for trouble anyway. First hop off my home lan over the wireless starts at about 9% packet loss. Sucks but for normal TCP based stuff (email, web, ssh, etc.) life goes on but just a little slower. 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 a whopping 29%. So between the last Internet router (bbnet2.lax.pnap.net) and broadvoice's edge router, there is an additional 16% loss. No wonder my outgoing voice to the PSTN is choppy, filled with several second gaps, and makes people laugh at me for spending $20 a month on VOIP. I admit I can help things a bit by getting an ADSL or SDSL link with a better provisioned uplink, but even if I had 0% loss to broadvoice, their own net connection seems seriously under-provisioned. One thing might be affecting this and make these numbers suspect--broadvoice might have QoS on the edge router such that non-RTP packets get lower-class status, so my UDP pings are artificially dropped in favor of real RTP traffic (actually, I'd be doing this if I were them.) Anyone care to comment on how realistic a test this is? I'll do these tests for a few hours and hit the different broadvoice proxy networks and see if there is a difference, and compare to loss rates for other sites over my ISP uplink. Anyway, my Digium 11b card comes in tomorrow, so I'll be off to more fun setting up the IVR and voicemail, etc., for my home line off the PSTN...love this Asterisk thing. -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
[Asterisk-Users] VoIP Provider problems
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. ___ 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
Re: [Asterisk-Users] VoIP Provider problems
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 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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
