Re: [Asterisk-Users] VoIP experiences with Cable and DSL
On Wednesday 04 August 2004 17:29, Steve Szmidt wrote: I'll end up with ADSL too so would you be willing to send me a copy of the config work you did? My rc.tc script is posted in the QoS with sveasoft thread, the archives should have it by the time you get this. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
On Tue, Aug 03, 2004 at 07:48:13PM -0700, Chris said: - Original Message - From: Steve Szmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 7:04 PM Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL Assuming you're talking about Random Early Detection, are you saying that all cable providers use it? No, not saying that, just that since it's so CPU friendly and can handle large bandwidth, it's an attractive choice... however because VoIP packets are so tiny and very latency sensitive, RED is their worst nightmare :( -Chris This is an interesting thread, but it's VERY difficult to follow when quoting is done incorrectly. If I had not read previous messages, I would not know who said what. I urge all outlook and outlook express users to install quotefix which fixes Outlook and OE's horribly broken behavior. http://home.in.tum.de/~jain/software/oe-quotefix/ and http://home.in.tum.de/~jain/software/outlook-quotefix/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
On Tue, 3 Aug 2004 12:07:14 -0400, Steve Szmidt [EMAIL PROTECTED] wrote: For a few years now I've operated with cable as the obvious choice, at least in my area where RoadRunner really built up a good network. It could be that for nation wide implementation VoIP really should be on DSL. (Unless of course you need a big pipe where a split T is the only higher option.) I currently use Cogeco cable in Oakville, ON, Canada. It has been fantastic! I don't think I've used a provider with as much available throughput (exactly as advertised). Only occasionally does the service go up and down, but that is infrequent. I have an external modem, and am using a pure VoIP setup with IAX trunking to my VoIP/PSTN gateway. Only occasionally do I get a dropped packet or something, but nothing to worry about. I spoke with my parents for an hour over the connection, and there was no problems (actually... I was getting some echo, but Asterisk nicely took care of it, and my parents were not aware of any echo cancelling going on until I told them what Asterisk was doing on my end, as I could hear it working). I will be using Cogeco again for me internet (cable) so that I don't have to pay Bell any money. Unfortunately my buzzer isn't going to work in the apartment, so I'll have to let guests in, but hey, I'll do a bit of leg work just to save any of my money going to the greater of two evils :) Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
Leif Madsen wrote: On Tue, 3 Aug 2004 12:07:14 -0400, Steve Szmidt [EMAIL PROTECTED] wrote: For a few years now I've operated with cable as the obvious choice, at least in my area where RoadRunner really built up a good network. It could be that for nation wide implementation VoIP really should be on DSL. (Unless of course you need a big pipe where a split T is the only higher option.) I believe this is a 'religious' discussion. I deployed a widespread (phoenix/california/hawaii) telecommuting setup for 50 employees using H.323 (not Asterisk - Altigen at the time). This was across probably 15 different providers networks and spread pretty equally between Cable modem/router and DSL. In all cases 'business class' services were ordered at the highest available speeds. The bottom line - after 2+ years we have had about equal amounts of trouble over both media types. When it's good it's just about perfect - when it's bad it's the same as bad cell phone connections. The bad times are infrequent on either media types. My .02$ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
On Tuesday 03 August 2004 21:16, Steve Szmidt wrote: I take it you paid $200 for the Sangoma? Yes I did, and it was the best $200 I ever spent on VOIP equipment. Relatively inexpensive and like I said it eliminated the guessing games and queueing garbage. Did you have to get through any hoops to get it up, or did it just autoconfigure, as advertized, and you were a happy camper? The autoconfigure did *not* work. Several little problems but all solveable. 1) 2.4.26 is not supported at the time of this email. It's coming, they say, but they have some other pressing issues with other equipment and customers. (2.4.26 is closer to 2.6.x in terms of some of the driver backend) 2) The autoconfigure says it'll compile in ADSL but it doesn't... I found I had to do a manual config and then specify all the protocols it said were default (ADSL among them) -- the screen looked as if I'd done an default install but I had to do it manually. That's it. After that it built, link went up without any hassle and it's been working great. Now, having said that, I've been having perfect audio for the past 3 weeks but this past week I am having choppy outgoing audio but my bandwidth consumption is well below the maximum so I'm trying to track down what changed. I don't believe it to be a problem with the sangoma card, though. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
[EMAIL PROTECTED] wrote: On Tue, 3 Aug 2004 12:07:14 -0400, Steve Szmidt [EMAIL PROTECTED] wrote: For a few years now I've operated with cable as the obvious choice, at least in my area where RoadRunner really built up a good network. It could be that for nation wide implementation VoIP really should be on DSL. (Unless of course you need a big pipe where a split T is the only higher option.) I currently use Cogeco cable in Oakville, ON, Canada. It has been fantastic! I don't think I've used a provider with as much available throughput (exactly as advertised). Only occasionally does the service go up and down, but that is infrequent. I have an external modem, and am using a pure VoIP setup with IAX trunking to my VoIP/PSTN gateway. Only occasionally do I get a dropped packet or something, but nothing to worry about. I spoke with my parents for an hour over the connection, and there was no problems (actually... I was getting some echo, but Asterisk nicely took care of it, and my parents were not aware of any echo cancelling going on until I told them what Asterisk was doing on my end, as I could hear it working). I will be using Cogeco again for me internet (cable) so that I don't have to pay Bell any money. Unfortunately my buzzer isn't going to work in the apartment, so I'll have to let guests in, but hey, I'll do a bit of leg work just to save any of my money going to the greater of two evils :) Leif Madsen. http://www.asteriskdocs.org I am using Shaw for cable access and Primus as my VSP. I find on a fairly regular basis calls will just drop after 10 minutes or so. The call doesn't hangup but you can't hear each other talk. I am not using any QOS or bandwidth rate control. Any ideas why this might be happening? Thanks, Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 04 August 2004 10:50 am, Andrew Kohlsmith wrote: That's it. After that it built, link went up without any hassle and it's been working great. Now, having said that, I've been having perfect audio for the past 3 weeks but this past week I am having choppy outgoing audio but my bandwidth consumption is well below the maximum so I'm trying to track down what changed. I don't believe it to be a problem with the sangoma card, though. Thanks, good notes to have! I was going to ask what kind of connection you have (cable or DSL) but then my mind caught up with me... and gave me a swift kick. I'll end up with ADSL too so would you be willing to send me a copy of the config work you did? - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBEVVBljK16xgETzkRAgOJAKDSKrLGeBDwlUYPzNzkkZSxwmQD6gCfV9B8 Yq+kadpiNSsP8oq6TrW74fs= =MjTS -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 04 August 2004 04:28 pm, Geoff Nordli wrote: I am using Shaw for cable access and Primus as my VSP. I find on a fairly regular basis calls will just drop after 10 minutes or so. The call doesn't hangup but you can't hear each other talk. I am not using any QOS or bandwidth rate control. Any ideas why this might be happening? One thing that happens with networks is they queue up data, at various points, because it comes in too fast to route straight through. While the data is sitting in that queue your VoIP equipment is not getting anything, or enough, so it goes silent. If that occurs for too long a time, or in both directions, it will terminate the call. A way around this is to put in some quality of service solution that can squelsh your out bound connection to a rate that can flow through onto the Internet backbone with needing to be buffered. (I use altq in OpenBSD.) Here's one way: Let's say you have 384k up. But your upline cannot handle all without buffering. So you drop back 50%, see if it works. If so you increase 25% (or original 384), and see if that's good. If not you cut back 12%, else increase another 12%. (Binary search method.) Repeat until you don't get queued. Of course it could be easier to just back off 10% at a time until it works too : ) Anyway, as soon as your volume is low enough not to be buffered you will have less dropped/silent connections. This is kind of funny because you are lowering your speed, but getting better quality. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBEVisljK16xgETzkRAjP0AJ9lOhst6lWQE2mfLL7e0BjD+XJEIACfc8vu UZJUSQFu809b6jOVt3EAujA= =CNkV -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
On Tuesday 03 August 2004 12:07, Steve Szmidt wrote: But with VoIP it has to go both ways and things like latency can easily become a big issue. (I have cable and it seems that I get sound degradations much easier than I'm comfortable with, yes it's a shared connection with occational POP traffic. Also, I'm only talking about dedicated network connections for final implementation.) As the old Rogers Cable and Bell HSE commercials used to slog it out with With cable you're all sharing a link, with HSE it's individual links -- there is some truth in that. You have a dedicated TX/RX interface with DSL; once you hit the DSLAM you are, of course, just part of some gigantic ATM flood but at least the bandwidth on that ATM network is likely far beyond what is normally available. With cable you're fighting to talk; something that QoS isn't going to help with in a CSMA/CD network. So, what I realized was that I have no real data to operate with is, and has anyone done an evaluation of typical needs which shows DSL better suited for VoIP? F.ex. cable shares the pipe and unless QoS is implemented can reasonably have more traffic issues than DSL. QoS isn't going to help you get to talk in a crowded CSMA/CD network. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
- Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 4:05 PM Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL On Tuesday 03 August 2004 12:07, Steve Szmidt wrote: But with VoIP it has to go both ways and things like latency can easily become a big issue. (I have cable and it seems that I get sound degradations much easier than I'm comfortable with, yes it's a shared connection with occational POP traffic. Also, I'm only talking about dedicated network connections for final implementation.) As the old Rogers Cable and Bell HSE commercials used to slog it out with With cable you're all sharing a link, with HSE it's individual links -- there is some truth in that. You have a dedicated TX/RX interface with DSL; once you hit the DSLAM you are, of course, just part of some gigantic ATM flood but at least the bandwidth on that ATM network is likely far beyond what is normally available. With cable you're fighting to talk; something that QoS isn't going to help with in a CSMA/CD network. So, what I realized was that I have no real data to operate with is, and has anyone done an evaluation of typical needs which shows DSL better suited for VoIP? F.ex. cable shares the pipe and unless QoS is implemented can reasonably have more traffic issues than DSL. QoS isn't going to help you get to talk in a crowded CSMA/CD network. -A. Being a cable user, the other thing I notice is that cable (or at the very least my ISP) also seems to suffer from ARP flooding... Billions and Billions of Are you there? Yes I am! Who Is at blah? I am at Blah! Crap every second, probably wasting like 512kbit of bandwidth just for DHCP and BOOTP crap... But for the most part I gotta say that the sustained transfer rates are WAY better than they ever were with DSL... And I don't notice too much difference in latency between the two... As the old Rogers Cable and Bell HSE commercials used to slog it out with With cable you're all sharing a link, with HSE it's individual links -- there is some truth in that. You guys probably remember the old ethernets where the ether was this long thick yellow cable (ThickNet) HFC is something like that, everyone is sharing the same link like with the old ThickNet and BNC networks, it is not switched at all until you get to the headend and as more people use the link, the more congested it becomes until it becomes unusable because even ARP messages can't go through... QoS isn't going to help you get to talk in a crowded CSMA/CD network. I might be misunderstanding you about QoS, but I know for a fact that it does help greatly because whether you use DSL or Cable, your bridge device (it's not a modem no matter how much people want to call it that, it's a bridge!) uses large buffered queues to achieve sustained transfer rates... this is awesome for bulk downloads but makes your VoIP conversation sound like you're on a cellphone under a bridge in a windstorm... Also if the ISP is using QoS and they classify users by the MAC address of your bridge device, they can create something similar to ATM PVCs, allowing traffic to flow more orderly and evenly across THEIR network... Bear in mind that when you're using QoS you're shaping YOUR traffic as it goes out YOUR link... you can do nothing about what happens to it once it crosses your ISP's router into the rest of the InterNet. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 experiences with Cable and DSL
On Tuesday 03 August 2004 19:44, Chris Shaw wrote: QoS isn't going to help you get to talk in a crowded CSMA/CD network. I might be misunderstanding you about QoS, but I know for a fact that it does help greatly because whether you use DSL or Cable, your bridge device (it's not a modem no matter how much people want to call it that, it's a bridge!) uses large buffered queues to achieve sustained transfer rates... this is awesome for bulk downloads but makes your VoIP conversation sound like you're on a cellphone under a bridge in a windstorm... Also if the ISP is using QoS and they classify users by the MAC address of your bridge device, they can create something similar to ATM PVCs, allowing traffic to flow more orderly and evenly across THEIR network... What I am saying is that you are shaping your ethernet to your cable modem (and yes I call it a cable MODEM -- you're still modulating and demodulating -- it's just DMT or some superhypermega modulation method) -- once it hits your cable modem you're playing the CSMA/CD game and if you collide you're SOL, there goes your timely packet. And yes I know all about huge queues... The cure for that (at least with DSL) is to get a Sangoma S518 -- it's a PCI ADSL modem with drivers for everything... I just prioritise packets now (no rate limiting) and get my full 4M/800kbit without any nonsense. I can flood the link in both directions and my VOIP sounds perfect. You just can't do that with an external modem -- tested 3 different ones (Speedstream one that comes with Bell HSE, an industrial grade one that comes with commercial DSL and also an old FP2100 -- the Bell one was by far the worst -- I had to rate limit to 400kbps or it would start queueing up the packets like crazy. Bear in mind that when you're using QoS you're shaping YOUR traffic as it goes out YOUR link... you can do nothing about what happens to it once it crosses your ISP's router into the rest of the InterNet. Exactly -- you're shaping your upstream and with a busy CSMA/CD or CA network you won't have much luck since your prioritised packets are getting delayed on their way to the head unit. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 03 August 2004 08:06 pm, Andrew Kohlsmith wrote: And yes I know all about huge queues... The cure for that (at least with DSL) is to get a Sangoma S518 -- it's a PCI ADSL modem with drivers for everything... I just prioritise packets now (no rate limiting) and get my full 4M/800kbit without any nonsense. I can flood the link in both directions and my VOIP sounds perfect. And cutting own my own speed to right below what I can push before they start queuing. The only problem with that, might be a fluctuating buffering level on their end as traffic changes. I take it you paid $200 for the Sangoma? Did you have to get through any hoops to get it up, or did it just autoconfigure, as advertized, and you were a happy camper? - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBEDjXljK16xgETzkRAkqxAKDZM9iu6v1Eyu2Va7rwokspHcUz/gCcDAU+ 4KJfRYIYVlBHQd2uLBVm7N0= =+cBv -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
- Original Message - From: Steve Szmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 6:16 PM Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 03 August 2004 08:06 pm, Andrew Kohlsmith wrote: And yes I know all about huge queues... The cure for that (at least with DSL) is to get a Sangoma S518 -- it's a PCI ADSL modem with drivers for everything... I just prioritise packets now (no rate limiting) and get my full 4M/800kbit without any nonsense. I can flood the link in both directions and my VOIP sounds perfect. And cutting own my own speed to right below what I can push before they start queuing. The only problem with that, might be a fluctuating buffering level on their end as traffic changes. I take it you paid $200 for the Sangoma? Did you have to get through any hoops to get it up, or did it just autoconfigure, as advertized, and you were a happy camper? - -- Steve The thing that really kills you on the ISP end is RED... it may be great for large traffic but it just KILLS voip... and there's not thing 1 you the customer can do about it... :( ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
- Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 5:06 PM Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL On Tuesday 03 August 2004 19:44, Chris Shaw wrote: QoS isn't going to help you get to talk in a crowded CSMA/CD network. I might be misunderstanding you about QoS, but I know for a fact that it does help greatly because whether you use DSL or Cable, your bridge device (it's not a modem no matter how much people want to call it that, it's a bridge!) uses large buffered queues to achieve sustained transfer rates... this is awesome for bulk downloads but makes your VoIP conversation sound like you're on a cellphone under a bridge in a windstorm... Also if the ISP is using QoS and they classify users by the MAC address of your bridge device, they can create something similar to ATM PVCs, allowing traffic to flow more orderly and evenly across THEIR network... What I am saying is that you are shaping your ethernet to your cable modem (and yes I call it a cable MODEM -- you're still modulating and demodulating -- it's just DMT or some superhypermega modulation method) -- once it hits your cable modem you're playing the CSMA/CD game and if you collide you're SOL, there goes your timely packet. And yes I know all about huge queues... The cure for that (at least with DSL) is to get a Sangoma S518 -- it's a PCI ADSL modem with drivers for everything... I just prioritise packets now (no rate limiting) and get my full 4M/800kbit without any nonsense. I can flood the link in both directions and my VOIP sounds perfect. You just can't do that with an external modem -- tested 3 different ones (Speedstream one that comes with Bell HSE, an industrial grade one that comes with commercial DSL and also an old FP2100 -- the Bell one was by far the worst -- I had to rate limit to 400kbps or it would start queueing up the packets like crazy. Bear in mind that when you're using QoS you're shaping YOUR traffic as it goes out YOUR link... you can do nothing about what happens to it once it crosses your ISP's router into the rest of the InterNet. Exactly -- you're shaping your upstream and with a busy CSMA/CD or CA network you won't have much luck since your prioritised packets are getting delayed on their way to the head unit. -A. Not really familiar with DOCSIS specs but I'm not sure cable IS actually CSMA/CD, it may be ATM or FDMA or even TDMA... I guess it depends on the provider? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
[EMAIL PROTECTED] (Chris) writes: The thing that really kills you on the ISP end is RED... it may be great for large traffic but it just KILLS voip... and there's not thing 1 you the customer can do about it... :( Interesting and somewhat disheartening. RED was really meant to put back-pressure on the protocols that understand a delicate touch, such as modern TCP. Trying to push back on UDP seems a bit pointless. I wonder if collective cry of the voip users can get the RED implementors to avoid whacking UDP packets until things get really dire (say when drop rates go past some magic number like 5% or 10%). -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
- Original Message - From: Steve Szmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 7:04 PM Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 03 August 2004 09:47 pm, Chris wrote: The thing that really kills you on the ISP end is RED... it may be great for large traffic but it just KILLS voip... and there's not thing 1 you the customer can do about it... :( Assuming you're talking about Random Early Detection, are you saying that all cable providers use it? No, not saying that, just that since it's so CPU friendly and can handle large bandwidth, it's an attractive choice... however because VoIP packets are so tiny and very latency sensitive, RED is their worst nightmare :( -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 experiences with Cable and DSL
- Original Message - From: Chris [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 7:48 PM Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL - Original Message - From: Steve Szmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 7:04 PM Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 03 August 2004 09:47 pm, Chris wrote: The thing that really kills you on the ISP end is RED... it may be great for large traffic but it just KILLS voip... and there's not thing 1 you the customer can do about it... :( Assuming you're talking about Random Early Detection, are you saying that all cable providers use it? No, not saying that, just that since it's so CPU friendly and can handle large bandwidth, it's an attractive choice... however because VoIP packets are so tiny and very latency sensitive, RED is their worst nightmare :( -Chris As the title of this thread is VoIP experiences with Cable and DSL... My experience with Cable and VoIP has been positive. I DO use an external cable modem with the infamous huge modem buffer and I find that I can set my QoS to about 80% of full speed and greatly reduce packet loss and jitter... So far when I ask people who I have talked to if they can tell it's coming from the internet or if it sounds like a cellphone they can't tell the difference... Mind you this is hardly a scientific test, and I do notice the occasional dropped packet, but for the most part it works damn good... So good in fact that I've dropped my telco completely and now have totally gone to a SIP-based setup with *, using a cellphone only for 911 calls if needed and in case the VoIP goes down... That's just my experience, I hope it helps you... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
