VoIP QOS best practices

Tue Feb 11 16:19:03 UTC 2003

>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
>  >
>  >