QoS on a Router / Switch

should we do something different?

Dunno. Too many variables and too many unknowns. It's often easier to measure your current VoIP performance than to calculate.

For 1 G.711 line, my 1.5Mbit/sec DSL line showed no data loss, but

10msec of jitter both ways, which is fairly good (MOS=4.0). However, when I tried the same test with 4 simulated lines, I got 8% packet loss, and 20msec of jitter, which sucks (MOS=1.0).

You will need to use QoS to prevent other traffic on your system from trashing your VoIP traffic. Since QoS is by IP socket number (ISO layer 3), it's all in the router. After enabling QoS, you should run the above tests again to make sure that you didn't reserve too much bandwidth for VoIP. Since you're running multiple VoIP services, you'll need to enable QoS rules for each one.

As a rather crude rule of thumb, you'll need 64KB of bandwidth for each user. So, if your 24/1Mbits/sec ADSL2+ connection to the internet, and you have 6 VoIP users, you'll need to reserve a maximum of 6 * 64 = 384Kbit/sec for VoIP. That should work. Fortunately, it's dynamic and it's unlikely that you'll have all 6 users on the phone simultaneously.

There are many such VoIP test servers scattered around the planet. If you plan to make international calls, you should also try using some on other continents.

--
Jeff Liebermann     jeffl@cruzio.com 
150 Felker St #D    http://www.LearnByDestroying.com 
Santa Cruz CA 95060 http://802.11junk.com 
Skype: JeffLiebermann     AE6KS    831-336-2558

andomly with laptops. There will also be the occasional visitor who may wi sh to connect wirelessly. This is not a call center sweatshop. Here's wha t I want:

have to worry about garbled audio when (perhaps?) network resources are in high demand.

etc... WTF?

Ethernet) with a 24MB U-Verse circuit. For the convenience of the occasio nal visitor or outside employee, we'll also deploy a Netgear N900 (WNDR4500 ). That said, none of the SIP phones or other permanent office gear (printe rs, etc..) will connect wirelessly - they will instead hardwire over to the aforementioned 24-port ProSafe Gigabit Ethernet switch.

or should we do something different?

existing VoIP server is fairly robust. And I don't mind over-provisioning the local circuit - I just don't want to have to deal with crap audio ever again.

Hello Sir, VOIP can be set up in a number of ways. A simple way would be to use Google Talk -- allows real-time conversation between US and Asia. Or one could use a Brekeke VOIP server and then use an softphone on any PC, configured to use that VOIP server. Skype is definitely an option, and does not require and Brekeke server configuration. So, one might start with a simple solution and then refine the system.

Newegg lists the switch is unmanaged, so I don't see how it is relevant. [I find netgear products to be flaky.]

QOS (well at least with DD-WRT can be a port range, ip range (net range, or a specific MAC address. It seems to me all you can do is spec a port range in your situation since all the ip addresses are essentially peers.

If you want to through some money at this, you can put all the SIP phones on their own lan, then have the router QOS that ip range (net range). I've eyeballed the dedicated lan for SIP in businesses, but don't know for a fact if they use QOS or just provide a separate line. [Like I'm going to track down the IT guy and ask.] With enough users, the lan segregation isn't that big of a deal, considering what employees cost.

what I want:

high demand.

etc... WTF?

switch.

You do realize that you are not answering the question that was asked.

Commodity routers have pre-defined service lists that include the necessary port numbers. There's no online emulator for the WNDR4500 on the Netgear site. Perhaps they'll post something after it's obsolete.

There's a series of screenshots at: The QoS enable/disable menu is shown, but the Rules page with the list of programs and services that appear when QoS is enabled is not shown: This is roughly what it looks like on a different Netgear router: Skype, SIP, and H.323 are all listed.

Give unto me a break. Most LAN's these daze are Gigabit (1000baseT) and will easily handle 6 VoIP lines at 80Kbits/sec maximum for G.711.

0.5Mbits/sec of VoIP traffic isn't going to put a dent in a 100 or 1000Mbit/sec LAN.

Most companies go to VLAN's before they segment their LAN's. Segmenting the LAN does nothing because the traffic on each segment in a star topology is just the traffic to and from a single workstation. It's the backhaul that gets overloaded with traffic. Switching to gigabit or fiber on the backhaul is the most common fix. The only separate LAN's I've seen and done were to keep the backup software off the user network. Backing up workstations over the network is usually done at night, when VoIP traffic is minimal.

--
Jeff Liebermann     jeffl@cruzio.com 
150 Felker St #D    http://www.LearnByDestroying.com 
Santa Cruz CA 95060 http://802.11junk.com 
Skype: JeffLiebermann     AE6KS    831-336-2558

Get a managed switch, this way you can adjust the speed per port. If you have multiple IP's available the you can split the network and with a good Qos router you can control the bandwidth in each segment.

Cheers

Hey, I only report what I see. Basically SIP phones with their own connection.

I haven't seen it on the left coast but I was in Wally World where all the phones were SIPs.