Very poor softphone voice quality even on LAN

I'm experimenting with using SIP and softphones on our LAN (and eventually WAN.) I'm amazed at how variable the quality is even on a lightly-loaded LAN. I have two test Windows 2000 machines connected to the same HP Procurve

4104GL switch (which is supposed to have QOS support enabled by default) at 100mbit with a softphone loaded on each. Conversation is great until one of the machines does almost any kind of network access. Then there is immediate "crunchiness" or skipping of audio. I've tried a variety of softphones (eyeP media, SoftJoy SJPhone) and codecs (G.729, G711ulaw) but all seem to have this problem. Am I missing something with configuring QOS with Windows 2000 Professional itself? I tried enabled 802.1p support on the NIC's but this didn't help either.

In message Mitel Lurker

802.1p/q has to be enabled throughout the LAN, not just on the NICs. Until

Sounds like you need to upgrade your switches (or stop using hubs?).

I some VoIP gear (3 Vonage lines, another hardware SIP device, softphones on a few machines, plus some audio streaming) across a 100Mb network.

All of the VoIP gear (Vonage and internal VoIP alike) are all connected to my primary network and there is no internal packet prioritizing taking place.

I regularly transfer data between servers and between workstations and two servers, bottlenecking at the 100Mb switch port connecting the server (And yes, a gigE upgrade is being budgeted, although only for the servers initially and one or two workstations -- The servers mirror large chunks of data between them though, so they'll benefit from the gigE even if the uplink to the network has to stay at 100Mb for the moment.

I've pushed 200-300Mb's through the switch (various machines transferring to other machines), plus saturated my upstream connection (to approximately 90% of it's capacity, which is 100% of my rate limiting -- The line is configured for 5Mb/1Mb), then picked up a Vonage line and made a crystal clear phone call.

The ONLY packet prioritization takes place on the border router / firewall at this time, the only thing I've done on the switch is to prioritize traffic from the firewall to the rest of the LAN. Since the WAN side of the firewall will only ever have up to 5Mb inbound, there is no risk of this traffic saturating internally.

On the firewall I prioritize ACKs, VoIP traffic, ICMP, DNS queries, HTTP+SMTP+IMAP user access, then all other traffic (in that order).

ICMP is only prioritized for testing purposes -- I can maintain 20ms ping times to my provider's first hop regardless of internal traffic, and my provider has sufficient bandwidth that nothing on their network is bottlenecking, so there is no need for QoS (As much as it would be nice, it's not affordable at this time)

All that being said, before I upgraded to a "real" switch I was using some cheapo SOHO Linksys gear and definitely noticed a performance hit on the Vonage equipment if my servers were performing nightly backups (to each other)

"Ed M" wrote in news:Exspd.5242$ snipped-for-privacy@news01.roc.ny:

You must have some problem on your network. If a single VoIP call (in any codec) doesn't have acceptable quality on such a simple setup, there's got to be a problem. Especially if the problems are comparable between G.711 and G.729. What does the CPU load on the PC's show?

That in itself could be a clue. Check the NIC card configuration on the PC to make absolutely sure you are configured (locked) for 100 megs and full duplex, not 10/100 and not half duplex and not auto-negotiate. Also make sure that nothing is connected to a HUB. Any HUB + VOIP = Misery.

What else is the PC doing? Can you arrange to halt all other PC activity and run the softphone application by itself?

Mitel's YA (Your Assistant) and YAPro softphones, at least in release 2.1 do not seem to have this problem. I'm running one on my AMD 2400 at home over a Linksys BEFSR41 router and VPN back to the Mitel 3300 VOIP pbx at the office and making both local and LD calls with no observed issues and with what I would certainly describe as "toll quality". The lone exception has been when listening to MOH you can hear occasional dropouts and scratchiness, but live voice is fine.

Thanks for all the replies to my question I'll answer the various questions and then see if anyone has any more ideas for me.

Yes but these are 100 mbit connections. I can see this happening on a WAN but on 100mbit LAN connections I would think that there would be plenty of bandwidth to spare.

I'm using HP Procurve switches. According to various reviews (Tolly Group etc.) these are as good as Catalysts for VOIP QOS purposes.

I'm beginning to wonder whether softphones can really deliver "toll quality" voice. Its seems that almost ANY activity on the PC's (from what I can tell even ones not involving network access) causes some audio degradation. I'm testing by making a call using the G711 codec (I've tried both U and A law) and putting myself on hold and listening to music. This codec is indistinguishable in quality from a POTS call (vs. G.729 which is nowhere near toll quality) but then there are the drop-outs. My test PC's are fairly high end (1+ ghz CPU's with plenty of RAM)

I then I wonder whether other customers of the softphones I've been trying expect toll quality voice. Is it possible that some people's expectations have gone down in this age of cell phones with their highly compressed audio and free VOIP where people are happy to have free calling even if it isn't perfect? The application I'm working on is for a sales department where the audio quality needs to be as close to toll quality as possible.

Sounds like you need better test tools.

has a list of them.

If you're doing this on a limited budget, try using a third machine to do the heavy network access. This will help you determine whether its a network issue or an issue with your local machine (e.g. not enough CPU to do both VoIP and the data stuff).

If it's a network issue, check out

and try the voice quality tests there.

Don't assume. Go look at the configuration and make sure. Also having QOS honored on the switch doesn't mean anything if the packets aren't tagged to begin with. Does your softphone software support QOS tagging or TOS tagging? The HP switch I believe will do QOS based on TOS tagging.

It's not just overall bandwidth. It's also latency which can be affected by bursts of traffic, both before it hits the NIC and after it hits the network.

I think I found the problem The three different Dell PC's I tested (two Dimension desktops and a Latitude laptop) must have very low-end sound cards built into them. Maybe they are software based? I tested using a Dell Optiplex using the same softphone on the same network and the sound was perfect.

Then I tested using a USB headset adaptor from GN Netcom (GN8110 USB) on the PC's that had trouble before and using that the quality is great. Apparently there is a DSP built into the cord that offloads the processing from the PC. So it looks like all along my problems were caused by the Softphones trying to use the low-end built-in sound cards. Strange that I see no other newsgroup postings about this or any warnings from the softphone vendors..... Everyone focuses on network QOS (which is very important but not the whole story.) Thanks for the input everyone!