Considering the "diagram" in the code snippet:
Is it possible to throttle the bandwidth on the SonicWALL (to say 1Mb?) and thus leave 500Kb always available to the VoIP network?
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Can you clarify for me, we're talking about the Pro1260? What release are you running? Standard or enhanced? Up to date support contract so you're getting updates? Have you tried SonicWall support (they are pretty darn good - fantastic in fact compared to other companies). Take a look at these articles. Cheers.
http://www.sonicwall.com/u
http://www.sonicwall.com/u
The Pro1260 is running Enhanced OS. We have a current contract and are fully up to date. I have tried SonicWALL support for other issues and met with nothing but frustration. That was quite a while ago -- maybe they have improved since then. I do think they make a fine product, and I even like their KB portal, but my last few experiences trying to work things out via phone have been less than positive.
Our VoIP is hosted and runs through a custom OpenBSD router. It works wonderfully! The only challenge is when we have heavy data traffic on the LAN. I've attached the SonicWALL document I followed to set up BWM. The SonicWALL and the VoIP network share a single T1. I have the SonicWALL throttled to 384Kbps up and 640Kbps down on the WAN connection. This should leave ~500Kbps for the VoIP network at all times. Yet, if I am on a phone call and then start a large download, I can immediately hear a loss of sound quality as the audio gets choppy.
What's still missing here?
Just come across SonicWall performance again...........seems to many question for really simple thing
http://www.experts-exchang
While I understand your point dkarpekin I don't think that is the issue here. (And the issue you just quoted is not really about BWM.)
I just got off the phone with SonicWALL support. Which I must say it was a MUCH better experience than previously!
They have pretty much confirmed (now having heard it from 3 technical people from 3 separate entities) what I have somewhat expected all along. The bottom line is that there is really no way to control the inflow of data through the T1 (at least not to any reasonable degree); other than if we could use some sort of CoS. But CoS is only valid if we can control the packets from start to finish -- which we can't in this case.
In the testing I did with SonicWALL they were able to hear me just fine, even with the network under heavy load. That's because we ARE able to control the amount of data leaving the T1 circuit via BWM on the SonicWLL. However, during heavy network use I was unable to hear them clearly due to the fact that, even though we were throttling the bandwidth on the SonicWALL, the T1 circuit was being more or less flooded on the inbound, causing the VoIP traffic to have to compete with the other traffic. (When I kick off a Linux distro download for example, it's like inviting millions to the party. Even though I control the door, my VIP guests (VoIP packets in this case) still have to fight the crowd just to get to the door.) A simple analogy perhaps, but I think it is basically accurate it relating what we are up against.
Test your peerformance after all , you can use PC-to-PC,across diffrent part of WAN or LAN:
Iperf 2.0.2 installer for Windows from
http://dast.nlanr.net/Proj
This will help narrow down problem, find "bottleneck",cable problem, diffrents in file transfer type, and so on.
Seperate IP's has nothing to do with the physical transport. Pretty much, sounds like you're hosed unless your carrier can lend you some assistance and apply a policy on their end, which I've never heard of. Do you have an option of bringing in another circuit for data? Perhaps a DSL or cable connection?
Unfortunately, we are pretty remote. We have a T1 due to some "special circumstances" and that works great. No DSL or cable however. I'd have already been all over that :-) Another T1 is an option, but more than we wanted to pay at this point. I guess it may come to that some day -- we'll see.
Thanks for all the input.
MTU settings ,might be helpfull to change on WAN interface according
http://www.dslreports.com/
http://www.dslreports.com/
It is save enouf to try 1300 for example..........most of the ISP does not go below, and try k-perf over WAN with apprpriate port(make sure no other service is using same ports) to see actuall diffrent in speed mesurements..........
k-perf also create tools to check particular ports performance due firewalls , all other degration contributors........
Sure, he can also uplug his WAN connection to see the effect of that, but it will not resolve his issue trying deal with a QoS issue.
Hope isn't a strategy and "Experts" shouldn't waste people's time with needless tests. His problem isn't fragmentation, and even if it was, whoopdeedoo, we've made his connection a little more efficient for what? Yep, let's sling those packets in smaller chunks a little bit faster!
You can control traffic by QoS, like 2800 series for example might suite you well.......
All details/advise you can easy verify with Cisco sales.
Also just pop in mine mind, not sure of yuor physical/VoIP structure, but if it is will be possible to assign VoIP gateway WAN IP and connect directly to ISP by any $30 4 port switch( if you nee one) bypassing stupid SonicWall.
For me not much problem "hanging" here , I'm borred at mine work........
The VoIP network is not going through the SonicWALL at all! They just share the same T1. That's why I was hoping to find a way to throttle the SonicWALL to 1Mb, so the VoIP router would have a GUARANTEED 500Kb at least for voice calls.
The problem is that I CAN control the outbound traffic from the data network, but i can NOT control the amount of data coming back in through the T1. Once the traffic is bunched up at the point of the T1 there is no way to guarantee throughput to the VoIP network. I hope I'm missing something here, but I don't see how a better router on our end of the T1 can fix this? And QoS is useless going from our VoIP host --> Internet --> AT&T network --> Our network.
Than I did not clealry undestand you diagram, sorry for that, putinig on drawing should help beter.
Whoever control T1 both side (ATT?), should be able priotarize VoIP traffic.
Also "incoming" traffic, on which you have no contol, might be restricted, and it can be wide variation how to do so.....
http://www.faqs.org/docs/L
I have talked with AT&T several times. According to them anyway (and I have talked to several people there -- including one network engineer) they cannot prioritize VoIP traffic coming from another network. Though I must confess I don't really understand why, here is the explanation they gave:
"How CoS works within AT&T. All traffic, regardless of what CoS method you use (IP, TCP Port, TOS, etc.) gets marked with DSCP. The DSCP gets read by MPLS of their network. From MPLS the markings get translated into EXP0 through EXP4. The crux of it all is& this only works within the AT&T network! All traffic coming from outside gets a CoS4 priority (the lowest priority.) In other words, AT&T only prioritizes the traffic within THEIR network. This is pretty much the case across the board. Or rather across ISPs. By design CoS requires very specific information written into the router tables. It would be EXTREAMLY unlikely that this would work across different networks (ISPs) unless those entities had specifically configured their routers to handle the traffic properly. Its one thing to prioritize the packets (and that will work as far as outbound traffic is concerned) but once the data hits routers that dont understand (or pay attention to) the markings then the prioritizing is rendered useless."
for what that's worth.
The link to the Squid HOW-TO is interesting, but I'm not sure I want to go down the proxy server road. (at least not yet...)
In this prospect, even if you'll be able achieve prioritization, it is still will downgraded by ATT network , any "spikes" on network will lead to choppy audio.
It is just strange to hear in this days, that such simple thing- like additional ISP can not be reachable.
I do think you need work closer with ATT, to find "way around", they do have fiber buried all over the country, and they hardly using it at full extend.
Thaat their strict policy makes people live miserable, but on other hand once you'll "reach out" their service reliability is top of the line.
You just need to have enough pation to get through its bureaucratic machine, getting dedicated VoIP line.
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by: dkarpekinPosted on 2008-09-04 at 18:14:46ID: 22394395
Sorry to say, but switching to cisco might be the solution............