Implemeting VoIP and QoS

Wireshark is the defacto tool for diagnosing VOIP problems.  It has a rather steep learning curve to get useful accurate information from it.  I think your problem is pretty basic in nature though.

1. What is your internet connection type and speed?  A DSL connection will never handle more than a couple trunks or remote extensions. 
2. What is the result of a VOIP speed test? 
3. How many SIP trunks? 
4. Who is the SIP provider? 
5. Ports are 5060-5062 TCP&UDP and your SIP provider needs to tell you what port range they want for RTP audio stream  10,000-20,000 UDP is just the starting point for a single trunk/phone.  (my 18 trunks run 10,000-60,000 UDP) 
6. Low end SOHO routers like the Buffalo and the Linksys WRTxx even with the public domain software replacements are simply inadequate to handling VOIP.  I have found that the SPI software will queue up packets on almost every SOHO router out there and will walk all over your QOS settings.  It takes a hardware firewall designed to optimize for VOIP or a really powerful PC running a software firewall to keep up with the RTP.  Consider a Draytek or Edgemark router witht eh SPI turned off they specialize isn small VOIP setups on Cable and DSL.. 
7. You can probably get a 40% improvement just by turning off the SPI. 
8. Softphones are notorious for chop and bad audio.  The QOS in WIndows is abismal.  Recommend >2GHz with a dual core processor, 2GB RAM of DDR2 or better and SATA drive.  Also analog headset on hardware audio jacks is preferred over a USB headset.  Do you experience perfect audio on extension to extension calls that are entirely on the LAN?  Don't perform CPU I/O intensive things while talking on the softphone as PC's NIC may have a little QOS control, the CPU and I/O bus are for the most part first come first served.  Burning an ISO on an IDE drive is not a good thing to do 
9. You need bandwidth management that is IP based and a fixed IP address for the asterisk box.  You need to reserve 100kbps of the upload speed for every trunk or remote extension on G711 CODEC.  You also need to leave an upload reserve for additional overhead generated by the asterisk box for DNS, and chat with the SIP registration and proxy servers. 

That;s enough of a list to get you some things to start working on.

Mark

Mark,
I appreciate your thorough reply. And it is a LOT more involved than I thought.  I think we do need to get more upload speed.  But just to narrow things down for this particular question ... the real problem is when I type in a URL, press "Enter" and my PC goes off to find the page, THAT'S when the quality breaks up (the person I am speaking with sounds very garbled to me).  It's more the stuff we are doing on our computers which we can control, rather than the public Internet which we cannot.  

You could say, just don't DO anything else while on a call, but that is not practical. We don't do anything in the background like processing video files or burning CDs or DVDs.  I have been using the USB interface but will try using the analog interfaces instead.

Should we maybe have a 2nd NIC card in each computer? Turns out there is another network in the office we rent independent of our DSL and we *could* websurf using that network, thus dedicating our DSL for the Asterisk system and our softphones only. Trouble is, I don't exactly know how to "segregate" computer activities whereby the softphone app would use 192.168.1.x and everything else on that computer (i.e. browsing) would utilize the other NIC card at 10.1.1.x.  Any ideas?  

Thanks again!

Steve

Ok, that's some useful info in your last post.  One of my clients is a call center that uses entirely softphones  for all the agents.  They make calls and are working in Excel spreadsheet call lists at the same time.  I am converting them to a web based database for calls.

In a nutshell, the PC workstation needs enough whoopie that it can handle the I/O bursts associated with the workloads of a call and the other activities.  When we set it all up we experimented with workstations extensively to avoid the specific problem you are outlining.

This is what we found:

A single core laptop (1.7GHz Centrino/1GBRAM) could barely handle a call and do nothing else.  It could not do it on the WIFI LAN but was better on hard wired LAN.

A desktop (2.2GHz E2xxx series dual core with 2GB RAM DDR2 (667MHZ) worked fine with a call and open spreadsheet and had minor audio degradation if they saved or opened.  If you opened a web page and it was loaded with other embeded sites or lots of graphics (Google/Yahoo/MSN)  audio was definitely degraded to be borderline unacceptable for business use but OK for SOHO/home use.

A desktop with 2.6GHz E2xxx series on a faster databus and 800Mhz RAM speed was much better but could be pushed into unacceptable audio if you had all the apps open on a typical business desktop while doing a call and did not curtail your normal activities.  Note this is under Windows XP and as the machines have aged and Microsoft Update has loaded in Service Packs and Security updates, particularly IE7, XP SP3 and Office 2007 SP2 it has become easier to overload the box.  These systems are for internal use only and have no Antivirus engine running - Norton or Kaspersky would push these systems over the edge.

Start here to isolate whether the real problem is PC capacity or networking (LAN/WAN/ISP).

1. It occurred to me to mention that you need to be on a wired LAN running on switches not hubs.  The Buffalo or any low end SOHO router with wireless capability will not work for RTP VOIP on its WIFI.  It just won't.   If you are running wireless, you need to fix that first.  
2. Make an intercom call extension to extension on the LAN and try on one machine only to open web pages, files, etc. and see what the audio quality is like then?  If audio is bad then you need to fix it at the PC's before you can proceed.  If the audio is perfect or acceptable, then it is time for the Buffalo to go. 
3. I am curious, what happens to your audio if you are on a call and someone on another PC on your LAN starts surfing web? 

Look at a Draytek 2910 or if you need wifi 2910g router.  For reference only, I sell a 2910 for around $175 but you can easliy find them for less from the big online retailers.  It can do the bandwidth management and prioritization for VOIP that you need.  It can also handle dual WAN as well if you go that route.  This particular router can fix a lot of problems in small VOIP systems.

Thanks again for the very thorough response.  

We do not have any hubs in our environment. I was told early on that hubs are VoIP killers and only switches should be used. When we frist had things set up, my associate had a hard-wired connection to the server and I used wireless -- it is a VERY small office, but I wanted to avoid a long wire running to the server.  I did away with that setup pretty quickly -- wireless VoIP was not a good idea and the quality was poor.

I have not done the internal intercom to intercom under heavy PC load test yet. I'll try that when I get back in the office next Tuesday. I will say that yes, the quality does break up for my colleague while I am futzing around on BitTorrent, so I bet that the Buffalo needs to go, right?

I saw the Draytek online for about $155 and may buy it. I still do want to have wireless and I do like the long reach of my Buffalo (and in my home, I have 2 Buffalo's WDS'd together, and need to implement better VoIP there as well).  If hang the Buffalo off of the Draytek, will I be messing up VoIP assuming I do all the traffc shaping with the Draytek?