RE: VoIP QOS best practices
But I could conceivably have 10+ voice channels over a T-1, I still don't quite understand how, without prioritizing voice traffic, the quality won't degrade... C. -----Original Message----- From: Bill Woodcock [mailto:woody@pch.net] Sent: Monday, February 10, 2003 1:20 PM To: Charles Youse Cc: nanog@nanog.org Subject: RE: VoIP QOS best practices > My main concern is that some of the sites that will be tied with > VoIP have only T-1 data connectivity, and I don't want a surge in > traffic to degrade the voice quality, or cause disconnections or > what-have-you. People are more accustomed to data networks going > down; voice networks going down will make people shout. It works fine on 64k connections, okay on many 9600bps connections. T1 is way more than is necessary. -Bill
> But I could conceivably have 10+ voice channels over a T-1, I still > don't quite understand how, without prioritizing voice traffic, the > quality won't degrade... Well, of course it all depends how much other traffic you're trying to get through simultaneously. Your T1 will carry ~170 simultaneous voice streams with no conflict, but you have to realize that they'll stomp on your simultaneous TCP data traffic. But you don't need to protect the _voice_... Look, just do it, and you'll see that there aren't any problems in this area. -Bill
On Mon, 10 Feb 2003 10:27:39 -0800 (PST), Bill Woodcock <woody@pch.net> said: Look, just do it, and you'll see that there aren't any problems in this area.
For those looking to "just do it", it's not very complicated or expensive to try -- and the quality is very, very good esp. if you have broadband. For an easy, step-by-step way to try it out over the public, non-QoS Internet, look at the steps at: http://www.pulver.com/fwd (yes, there are other free, public SIP servers, but I haven't found any with as much useful "documentation" and I'm not associated with pulver.com except for being an enthusiastic FWD user) FWIW, I purchased a Cisco ATA-186 and then a 7960 on eBay (after trying out MS Messenger and finding it lacking) and they just work. I also have used the same units to get a PSTN phone number routed over IP using www.iconnecthere.com -- and you can make it work behind NAT too (but I can assure you it's easier without NAT). I'm willing to play tech support via email if anyone has questions about getting started. Adi
On Mon, 10 Feb 2003, Aditya wrote:
FWIW, I purchased a Cisco ATA-186 and then a 7960 on eBay (after trying out MS Messenger and finding it lacking) and they just work. I also have used the same units to get a PSTN phone number routed over IP using www.iconnecthere.com -- and you can make it work behind NAT too (but I can assure you it's easier without NAT).
Vonage (vonage.com) let's you get your feet wet at $25/month. Limited outbound, but unlimited inbound and you can pick from many area codes. They supply the ATA, and you have 30 days to play. IConnectHere.com is the consumer arm of Delta3. They are OK, but they offer no help if you get stuck. Vonage is truly plug-n-play. Works fine behind NAT, doesn't require any ports to be opened to function behind a nat or firewall. Just make sure 5060/udp and 69/udp can go out and you're off and running. As others have stated, it's more fun to talk about VoIP after you've used it. I've found the voice quality equals or exceeds my POTS line. There is some echo at times when the call starts, then the magic echo-cancellation stuff seems to learn and things get better. The delay is fine, but can be a bit off-putting during a multi-person conference call between excited tech and marketing folks. But if you regularly use a cell phone, you may not even notice this, as I find the delay on my cell to be worse. What I'm guessing Bill is getting at is the common VoIP implementations out there are running UDP. Since it's in "spray and pray" mode, you'll be worried more about it stepping on your well-behaved TCP traffic than vice-versa. I'm running a codec that tops out around 80Kb/s on an ADSL line and I've yet to find a way to affect my voice traffic. In 6 months of using the service I've yet to have a dropped call, and I regularly make 80 minute+ calls. All in all I think there's less voodoo involved than most people imagine. It just works. Now I need to figure out how to break into my ATA so I can use it for FWD as well (the ATA ships with an md5 key and the config it fetches via tftp is encrypted)... Anyone? C
I'm willing to play tech support via email if anyone has questions about getting started.
Adi
On Mon, 10 Feb 2003, Aditya wrote:
FWIW, I purchased a Cisco ATA-186 and then a 7960 on eBay (after trying out MS Messenger and finding it lacking) and they just work. I also have used the same units to get a PSTN phone number routed over IP using www.iconnecthere.com -- and you can make it work behind NAT too (but I can assure you it's easier without NAT).
Vonage (vonage.com) let's you get your feet wet at $25/month. Limited outbound, but unlimited inbound and you can pick from many area codes. They supply the ATA, and you have 30 days to play.
IConnectHere.com is the consumer arm of Delta3. They are OK, but they offer no help if you get stuck. Vonage is truly plug-n-play. Works fine behind NAT, doesn't require any ports to be opened to function behind a nat or firewall. Just make sure 5060/udp and 69/udp can go out and you're off and running.
As others have stated, it's more fun to talk about VoIP after you've used it. I've found the voice quality equals or exceeds my POTS line. There is some echo at times when the call starts, then the magic echo-cancellation stuff seems to learn and things get better. The delay is fine, but can be a bit off-putting during a multi-person conference call between excited tech and marketing folks. But if you regularly use a cell phone, you may not even notice this, as I find the delay on my cell to be worse.
What I'm guessing Bill is getting at is the common VoIP implementations out there are running UDP. Since it's in "spray and pray" mode, you'll be worried more about it stepping on your well-behaved TCP traffic than vice-versa. I'm running a codec that tops out around 80Kb/s on an ADSL line and I've yet to find a way to affect my voice traffic. In 6 months of using the service I've yet to have a dropped call, and I regularly make 80 minute+ calls.
All in all I think there's less voodoo involved than most people imagine. It just works.
Now I need to figure out how to break into my ATA so I can use it for FWD as well (the ATA ships with an md5 key and the config it fetches via tftp is encrypted)... Anyone?
Tough one there. I've tried, but the only thing I've been able to do is reset to factory defaults. In any case, the current ATA software (2.15) doesn't support multiple proxies; you can have two accounts, but they seem to only use one gateway/proxy (and a failover.) Any evidence to the contrary is welcome. I found the way around this is to use Asterisk (http://www.asterisk.org/) and register my iconnecthere.com account from the server. I can have as many SIP accounts registered at the server, and they all act as incoming "channels" that can then be routed to my ATA-186 (or to voicemail, or to an IVR, or whatever.) I've had success in the last two days in getting my analog line at the house, my INOC-DBA phone, my iconnecthere.com account, and a SIP gateway on the other side of the continent to all make calls inbound/outbound from my single ATA-186 on my desk. There are still some bugs to be worked out, but it's rapidly getting to be a locally-controlled voice system for multiple gateways. FWIW, I'll be posting a summary on the INOC-DBA list shortly on how to get it working. Now, back to the NANOG-ish content: I know a fundamental change in technology when I see it, and VOIP is an obvious winner. VOIP has been smoldering for a few years, and the sudden growth of various easy-to-implement SIP proxies and service platforms, plus the sudden drop in price of SIP hard-phones, is going to push growth tremendously. Currently, the underlying technology is UDP that moves calls around. This is all well and good until you get thousands, tens of thousands, hundreds of thousands of calls going at once. QoS is, as Bill says, not a problem right now on public networks; I've used VOIP across at least three exchange or peering sessions (in each direction, no less!) and suffered no quality loss, even at 80kbps rates. However, when a significant percentage of cable and DSL customers across the country figure this technology out, does this cause problems for those providers? Is it worthwhile for large end-user aggregators to start figuring out how they are going to offer this service locally on their own networks in order to save on transit traffic to other peers/providers? Or is this merely a tiny bump in traffic, not worth worrying about? More interestingly: what happens to the network when the first "shared" LD software comes into creation? Imagine 1/3 (to pick a worst-case percentage) of your customers producing and consuming (possibly) 80kbps of traffic for 5 hours a day as they offer their local analog lines to anyone who wants to make local calls to that calling area. Overseas calling I expect will show similar growth. Nobody wants to pay $.20 or even $.10 per minute to Asian nations, so as soon as Joe User figures out how this VOIP stuff works, there will be (is?) a tendency for UDP increases on inter-continental spans. Nothing new here; we've all said this was coming for years. Now it's finally possible - is everyone ready? JT
C
I'm willing to play tech support via email if anyone has questions about getting started.
Adi
participants (5)
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Aditya -
Bill Woodcock -
Charles Sprickman -
Charles Youse -
John Todd