Bandwidth Management to address delays?

[corchard]

I'm new to this whole technology, but I am technical.

I live in Canada, and am getting .5-1 second delays when speaking. There is no echo, so you end up talking over each other.

I am on a high bandwidth cable (Shaw Internet) but have several other systems and applications that sporadically use the pipe and wonder if a bandwidth manager will help. Moreover if anyone knows of a suitable manager if at all for use with the MagicJack.

your thoughts are appreciated.

Chris

[HolmanGT]

corchard,

I would think a manager would NOT help much and the delay is most likely not on your end but some server you are going thru. Do you have this problem all the time?

The MJ uses such a small amount of bandwidth ... I think about 80 kHz which is like nothing. The only way in my humble opinion that it could be on your end if whatever it is that you say kicks on from time to time is a real bandwidth hog.

Frankly I have done things like burn CDs, play music and surf the WEB with no ill effects. I have also used Remote assistance to my Daughter and MJ to provide Voice communications while we trouble-shot whatever she has recently broken. Now that has to be a heavy load on my system which only has 512 up and have never experienced any noticeable delay.

I don't know what the problem could be because even a slow server is not going to be in your path all the time, unless your ISP has chosen to intentionally do this like some of them actually block VOIP for whatever reasons known only to them. You could ask them... who knows maybe they will tell you the truth.

[SpamBox]

I'm new to this whole technology, but I am technical.
I live in Canada, and am getting .5-1 second delays when speaking. There is no echo, so you end up talking over each other.
I am on a high bandwidth cable (Shaw Internet) but have several other systems and applications that sporadically use the pipe and wonder if a bandwidth manager will help. Moreover if anyone knows of a suitable manager if at all for use with the MagicJack.
your thoughts are appreciated.
Chris

Those of us who are using QOS, swear by it. So i would recommend it - can't hurt, but will probably help.

[testing123]

I'm new to this whole technology, but I am technical.
I live in Canada, and am getting .5-1 second delays when speaking. There is no echo, so you end up talking over each other.
I am on a high bandwidth cable (Shaw Internet) but have several other systems and applications that sporadically use the pipe and wonder if a bandwidth manager will help. Moreover if anyone knows of a suitable manager if at all for use with the MagicJack.
your thoughts are appreciated.
Chris

Those of us who are using QOS, swear by it. So i would recommend it - can't hurt, but will probably help.

I concur.

corchard: You have to do the best you can to reduce latency within your own private LAN.

1. This is best accomplished with QoS -- find out if your Router has it. Check here. It should help tremendously. If your Router supports DD-WRT, use it. NOTE: This may invalidate your warranty. Install at your own discretion and risk.

2. Understand what your bandwidth actually is. Visit this site (yes, they are US test sites, sorry)
This may help because many QoS need to know this.

3. Check your ping times to MJ. In Windoze:
Start / Run / CMD (opens a DOS window)
_tracert proxy1.Newark.talk4free.com (this is an EXAMPLE, depends on which MJ Proxy you are contacting)
^ (remove the "_" above because the Forum won't allow me to post that word)

4. What are you looking for?
a. # of Hops (less is better; 11 - 13 are typical)
b. Ping Times (less is better)
c. Dropped Packets ("*" is bad)

Problem is, if you have these there is not much YOU can do (except help identify WHERE the issue is)

5. Also, are you using DSL?? If so, you need to make sure you are using the optimal MTU size on your router. In the "CMD" window, do this:

ping proxy1.Newark.talk4free.com –f –l 1500
(that’s a lowercase “L”)

1) It doesn’t matter if you get a ping response from wherever you are pinging.
2) Pay attention to if you get a “Packet needs to be fragmented but DF set.” message
3) If you get that message lower the number (by 10 or 12 at a time) until the message goes away.
4) When it goes away, increase the number by 1 or 2 to determine your MAX packet size that does not have to be fragmented
5) When you find your MAX, add 28 to your MAX (20 bytes for IP Header, and 8 bytes for ICMP Echo request header). You MUST add the 28 to #4

They typically recommend a MAX of 1472 for DSL connections. You can figure out how to change the setting on your Router using the router link above.


A little geeky, but it may help.

[SpamBox]

WOW great post testing123!

Make this a STICKY!!!

[testing123]

WOW great post testing123!

Make this a STICKY!!!

Thanks, just doing my part to offer free tech support on MJ's behalf. LOL

[mufon]

Don't forget to set max mtu in Windows, or you will still frag packets. It's usually more effective here since it's an endpoint. Windows supposedly has a detect path mtu feature implemented in the ip stack, and has to be set in the registry. Even so it doesn't work (remember we're talking about Windows) , so set the mtu manually in Windows after you determine the correct value, and should be the same or less than your router's mtu. Ping from your router if you can, then do the same from windows to get the correct values for both. Usually they (mtu) will be the same for both but not always.

[testing123]

Don't forget to set max mtu in Windows, or you will still frag packets. It's usually more effective here since it's an endpoint. Windows supposedly has a detect path mtu feature implemented in the ip stack, and has to be set in the registry. Even so it doesn't work (remember we're talking about Windows) , so set the mtu manually in Windows after you determine the correct value, and should be the same or less than your router's mtu. Ping from your router if you can, then do the same from windows to get the correct values for both. Usually they (mtu) will be the same for both but not always.

Mufon, you bring up an interesting point -- do you REALLY have to setup MTU size on the Internal network when it is going thru a Router?? Does it not rewrite TCP packets as part of the NAT process?

I am not so sure. I think that if you Router has MTU size setting you may not have to do this on your Internal network devices (I mean it REALLy would be a pain to have to do that on every PC ...)

PS: If you DO want to change MTU size on your individual PC, you can try this program (DrTCP)

[LikeMagic]

Routers and PCs which must "talk" TCP/IP must have their "own" MTU value defined.

The router could have defined its own value (such as 1492 or 1500) because it must "talk" to the ISP with optimum MTU value to minimize overhead thus more efficient data transfer. Most DSL/PPPoE ISPs prefer MTU=1492 and most cable ISPs prefer MTU=1500.

You should also define the optimum MTU value for each of your PC's network connections (built-in LAN connection, built-in WLAN connection, PC card WLAN connection, etc) to match the MTU setting of your router to get the best optimized throughput. Windows' TCP/IP process will then use the network connection's defined MTU setting to "fill up" each TCP/IP packet for transmission. If you don't define the MTU value for each network connection that your PC has, then the OS may assign a lower default value such as 576 which may not be efficient for TCP/IP transmission.

Basically, if you define the optimum MTU value for your router, then the router will have optimized transmission to/from the ISP. If you define the optimum MTU value for each of your PC's network connection, then the PC will have optimized transmission to/from the router. Make sense? Yes?

[testing123]

Routers and PCs which must "talk" TCP/IP must have their "own" MTU value defined.

The router could have defined its own value (such as 1492 or 1500) because it must "talk" to the ISP with optimum MTU value to minimize overhead thus more efficient data transfer. Most DSL/PPPoE ISPs prefer MTU=1492 and most cable ISPs prefer MTU=1500.

You should also define the optimum MTU value for each of your PC's network connections (built-in LAN connection, built-in WLAN connection, PC card WLAN connection, etc) to match the MTU setting of your router to get the best optimized throughput. Windows' TCP/IP process will then use the network connection's defined MTU setting to "fill up" each TCP/IP packet for transmission. If you don't define the MTU value for each network connection that your PC has, then the OS may assign a lower default value such as 576 which may not be efficient for TCP/IP transmission.

Basically, if you define the optimum MTU value for your router, then the router will have optimized transmission to/from the ISP. If you define the optimum MTU value for each of your PC's network connection, then the PC will have optimized transmission to/from the router. Make sense? Yes?

LikeMagic, now this is interesting; again you raise excellent points. This technique (modifying MTU size) basically gained popularity during Dialup days when an individual PC had to connect to the 'net ... no router or other PC's at home connected.

The challenge is that I think this can (quickly) become a slippery slope that is beyond the avg User. Here is what I mean:

1) This link on DSL Reports "How do I optimize my MTU and RWIN settings" agrees with you and says EVERY device on the LAN has to share the MTU size.

2) This is another script on DSL Reports "How do I find the optimum MTU setting that basically is a fancier version of what I posted earlier (with screenshots! ). Of course, they say make all of them the same (same info source as #1 above)

3) Look here "VOIP and MTU size to stop jitter. Basic conclusion: (paraphrasing big time here) fix fragmentation by giving VOIP its own NIC on your Router and settings its MTU size (for the VOIP traffic).

4) If you are a Gamer or use VPN, the optimal MTU size for those apps may be different than VOIP. See here regarding Xbox, for example.

The theme I see coming up is that yes, all devices in a network should have a matching MTU size for optimal performance BUT the optimal MTU size for VOIP applications may differ/conflict/interfere w/ other types of applications (i.e. VPN, Gaming, etc.) on your network.

The ultimate approach (again beyond what the typical User has setup at home) is to have the VOIP "network" communicate w/ the Router on a separate NIC than the other network applications. (For example, you could accomplish this if you were running a multi-NIC router such as M0n0wall or pfSense on a soekris or on a PC w/ multiple NICs ... but not too many are willing to go to these lengths.

So, which MTU will be the best? Methinks it will depend on what app is most important to the User unless they can have separate NICs handling separate application categories. (And obviously, someone experience VOIP jitter who wants to fix it will probably think that most important and setup all the PCs to match the MTU size that they find to address that problem; unless it messes up their gameplaying LOL)

[Gerry_52]

Using logic...which according to Mr. Spock is essential...if you change the MTU to say 200 from what is possibly now in the area of 1500 to reduce stutter, wouldn't you now need to send roughly 7 times more packets to accomplish the same thing. Over time, epecially if your upload speeds are marginal couldn't this increase lag times? Since I use a Sprint wireless card for my internet I at times do have problems with stutter. I average about 950kbps down and 225kbps up, however during peak times the upload speeds can drop to 75kbps, which is at best marginal for voice communications. The MJ on a laptop with a wireless card is an elegant solution when it works and for a RF engineer is just geeky enough. I am working on educating myself on terrestrial based landline communications, but the more I read the more I realize I have a lot to learn. I am thoroughly enjoying learning however, and the communications periodicals I digest every month are telling me that the money is going to be in engineering, installing and servicing commercial VOIP services. Anyways, I'll help where I can, and the rest of the time I'll listen and learn from you guys.

[testing123]

Using logic...which according to Mr. Spock is essential...if you change the MTU to say 200 from what is possibly now in the area of 1500 to reduce stutter, wouldn't you now need to send roughly 7 times more packets to accomplish the same thing. Over time, epecially if your upload speeds are marginal couldn't this increase lag times? Since I use a Sprint wireless card for my internet I at times do have problems with stutter. I average about 950kbps down and 225kbps up, however during peak times the upload speeds can drop to 75kbps, which is at best marginal for voice communications. The MJ on a laptop with a wireless card is an elegant solution when it works and for a RF engineer is just geeky enough. I am working on educating myself on terrestrial based landline communications, but the more I read the more I realize I have a lot to learn. I am thoroughly enjoying learning however, and the communications periodicals I digest every month are telling me that the money is going to be in engineering, installing and servicing commercial VOIP services. Anyways, I'll help where I can, and the rest of the time I'll listen and learn from you guys.

Well Gerry_52, you know everything in life has its tradeoffs, huh? And I am definitely not even trying to suggest something as low as 200. The more typical (from what I have seen, especially for DSL) is 1300 - 1472.

But yes, smaller packets will require more packets to transmit the same data; I believe one of the links posted above actually talks about that a bit (maybe the Xbox one I think).

Yes, learning this stuff is pretty cool and VOIP may very well be the next hot thing.

[Gerry_52]

I'm just wondering how much effect setting the MTU on your machine is really going to matter in the big picture. Won't the packet size be dynamically changed several times downstream before it reaches it's destination anyways, or is preferable if not important to set the packet size prior to introduction to the internet to prevent excessive fragmentation once the packet hits the IP layer?

[HolmanGT]

testing123,

And with the hypothetical 200 for an MTU you might actually be Transmitting more packet overhead than data. I don't remember the exact figures but I read on DSLreports that IP packets at 1472 - 1500 are about 12% overhead (none data). At an MTU of 200 the data to packet overhead would be insane.

[Gerry_52]

At that point packet loss becomes catastrophic because I would imagine a fair portion of that information keeps the call in session. The other problem, if you want to call it that is I am running a Kyocera KR1 router. It's primary feature is that it accepts the Sprint PCMCIA wireless card and gives me both WAN and WLAN (G)...however it is not a very feature rich router as settings go. There is no way for instance to set QoS, but still...it is just way cool for what it does do. I however have not found much of a difference in QoS between running the MJ plugged into my desktop which runs on a home network thru the Kyocera with the internet accessed via the Sprint card and the Sprint card plugged directly into my laptop via PCMCIA and the MJ plugged into a USB port on the laptop. The results of side by speed measurements are also very similar taking into consideration the constantly changing RF environment wireless internet lives within.

[mufon]

Realistically, the endpoints are what really count, because that is where packets gain payload. The mtu at your endpoints have to be equal to or smaller than the smallest mtu of the path between endpoints. The optimal mtu is always going to be the mtu that fits the pipeline between endpoints without splitting the application layer's payload, therefore your mtu at your pc should equal the lowest mtu along the path between endpoints. Anything lower than that is not necessary unless you are unsure of the path mtu. In that case a little too low is better that a little too high., but going is low as 200 is just ridiculous and would only increase the aggregate latency of payload delivery, unless of course your path mtu is 200. Not likely unless your using dialup or pony express. If someone has a "stutter"problem they may want to see a therapist, but lowering endpoint mtu below path mtu isn't going to be of any help. Jitter on the other hand is not an mtu problem but rather is because of packets that arrive out of sequence and the transport protocol is udp. Since jack's jitter buffer is set in the registry and changes made will get overwritten the next time jack starts, then the next option is to set qos, tos, or cos if possible. Start by adding qos to your ip stack in Windows, and set any settings your router may provide. Sip should be "cs3" and rtp should be "ef". Simple qos settings would be priorities 3 and 5 respectively, or "maximize reliability" and "minimize delay" respectively. Tos values of 0x68 and 0xB8 are good if appropriate.for your router. Settings that maximize throughput or bandwith are tempting but won't provide the result you seek.

[testing123]

testing123,

And with the hypothetical 200 for an MTU you might actually be Transmitting more packet overhead than data. I don't remember the exact figures but I read on DSLreports that IP packets at 1472 - 1500 are about 12% overhead (none data). At an MTU of 200 the data to packet overhead would be insane.

That is correct ... I never suggested 200 -- that is ridiculously low. Typical that I have seen, (on a DSL connection, for example) is 1300-1472.

At that point packet loss becomes catastrophic because I would imagine a fair portion of that information keeps the call in session. The other problem, if you want to call it that is I am running a Kyocera KR1 router. It's primary feature is that it accepts the Sprint PCMCIA wireless card and gives me both WAN and WLAN (G)...however it is not a very feature rich router as settings go. There is no way for instance to set QoS, but still...it is just way cool for what it does do. I however have not found much of a difference in QoS between running the MJ plugged into my desktop which runs on a home network thru the Kyocera with the internet accessed via the Sprint card and the Sprint card plugged directly into my laptop via PCMCIA and the MJ plugged into a USB port on the laptop. The results of side by speed measurements are also very similar taking into consideration the constantly changing RF environment wireless internet lives within.

And that is entirely possible ... jitter is due to packets arriving out of sequence which MAY be helped considerably by QoS (i.e., in particular when you have multiple apps using your upstream bandwidth and "clogging the pipe"). Prioritizing VOIP packets over other network traffic can help to ensure that those packets arrive in sequence and w/o unnecessary delay. We all are simply trying to do the best we can to those packets while the packets are in our Network and on their way to our ISP. Beyond that physical point, there isn't a lot we can do.

Realistically, the endpoints are what really count, because that is where packets gain payload. The mtu at your endpoints have to be equal to or smaller than the smallest mtu of the path between endpoints. The optimal mtu is always going to be the mtu that fits the pipeline between endpoints without splitting the application layer's payload, therefore your mtu at your pc should equal the lowest mtu along the path between endpoints. Anything lower than that is not necessary unless you are unsure of the path mtu. In that case a little too low is better that a little too high., but going is low as 200 is just ridiculous and would only increase the aggregate latency of payload delivery, unless of course your path mtu is 200. Not likely unless your using dialup or pony express. If someone has a "stutter"problem they may want to see a therapist, but lowering endpoint mtu below path mtu isn't going to be of any help. Jitter on the other hand is not an mtu problem but rather is because of packets that arrive out of sequence and the transport protocol is udp. Since jack's jitter buffer is set in the registry and changes made will get overwritten the next time jack starts, then the next option is to set qos, tos, or cos if possible. Start by adding qos to your ip stack in Windows, and set any settings your router may provide. Sip should be "cs3" and rtp should be "ef". Simple qos settings would be priorities 3 and 5 respectively, or "maximize reliability" and "minimize delay" respectively. Tos values of 0x68 and 0xB8 are good if appropriate.for your router. Settings that maximize throughput or bandwith are tempting but won't provide the result you seek.

Bingo. Problem is that most consumer-grade routers will not have this level of control (i.e., "maximize reliability" or "minimize delay"). Typically just "higher" or "highest" priority vs. normal vs. low.

[Image]

Anyone care to give specifics on how they configured QoS for MagicJack?

In reading this sticky's title, I know that's what I was hoping for and not an MTU debate!

[HolmanGT]

Image,

I'll see your bet and raise you one. I don't even know where you go to configure the QoS.

I did turn the one in my router off because I made a completely hypothetical assumption that you don't want two QoSes running on the same system.

So:
1. Router QoS = off.
2. Windows Vista Ultimate QoS = on and un-tuned (don't know how, didn't know you could).

[dbarber]

Image,

I did turn the one in my router off because I made a completely hypothetical assumption that you don't want two QoSes running on the same system.

So:
1. Router QoS = off.
2. Windows Vista Ultimate QoS = on and un-tuned (don't know how, didn't know you could).

I would think that you would want the opposite. Have the router QoS on so that you could control network bandwidth utilization. That way you could prioritize packets from your MJ machine over any other traffic. If there are no other computers on your network......., NEVER MIND!

Donn

[saxman]

I have dsl-I ran the voip test last week and it was not good. I installed the Hawking HBB1 Broadband Booster-The new test shows a perfect score and the sound is great! It's worth the 25 dollars!

[Gerry_52]

I have dsl-I ran the voip test last week and it was not good. I installed the Hawking HBB1 Broadband Booster-The new test shows a perfect score and the sound is great! It's worth the 25 dollars!

Where did you find it for $25.00? I'm wondering if this will help with VOIP using a Sprint Wireless card? The wireless card is plugged into a Kyocera router which has no QoS settings available. Might be worth a shot for $25.00.

[saxman]

I got mine on sale at Fry's Electronics. I don't know what all it helps but that sounds like it would. I works to improve the connection speed and my dsl speed has definetly improved as well as the voip quality-Check it out here
http://www.hawkingtech.com/products/pro ... ProdID=216

[msiam]

My sister in the sticks of FL needs something like that for her Verizon Wireless card. I cant seem to get her a good conversation out of her MJ.. I'm in WI so I cant help from here.. Please let me know how you would set that up with her configuration of the wireless card. Gerry, do you have a wireless sprint connected somehow to a router? How did you do that?? I am curious to the scenario and configuration on that. What router did you use so far? Sis needs it!!

[Gerry_52]

I am using a Kyocera KR1 router. It accepts a PCMCIA wireless card and gives you both wired (4 RJ45 jacks) and 802.11g wireless. It is very cool. It also runs on 12V and comes with both a cigarette lighter plug and a wall wart. I am involved in emergency communications, so when I put it in the vehicle it gives me a rolling hotspot for all intents and purposes. I have used both Vonage and the magicjack with it. I'm not sure why but the Vonage phone will generally outperform the MJ. I'm hoping that with the release of the new codec for the mj the performance will improve. I am also a T Mobile subscriber and with the new Curve from Blackberry and the $10 a month phone@home plan you can use your cell phone over wi-fi to make unlimited calls for the $10 a month. So, if you take the Kyocera router with you and you initiate the phone call on the Kyocera using wi-fi and transition to the cell system by walking away from your car or turning off the router the remainder of the call will not be charged against your minutes...so how cool is that!

[Saratogamk5648]

For XP, Cablenut works well in setting all values for DSL, download it and the Upgrade settings. DSL normally has a 1492 DSL value, the RWIN also very important (Cablenut will set this according to your computer memory, speed and latency calculations). Doesn't work with Vista (what does!). There is a small application called WLAN Optimizer .NET that shuts down the WLAN Zero Config service. Speaking delays sometimes occur when VISTA auto searches for more optimum connections and can affect your connection whether you're using WLAN or wired. QOS, am using it but am not convinced of it's need, bandwith needs with MJ is just too small.

[BigRick]

The MJ uses such a small amount of bandwidth

I believe this is the problem, insuficiant bandwidth for MJ to utilize.

[dontazzz]

I have the same thing going on and mj is trying to help....and I am trying to read here and I am not very good with computers so thought the one two three would work but it getting to be more work. I wish it was easy cause right now my head is hurting and I have try three different phones thinking it was the phone and the (mj chat) person said it does not work with wireless have to be hooked up to the router. so I wrote to ask for a return number and (mj chat)another person has try to fix it and we went deeeper in to my computer and then down loaded divx which I have not figure what that has to do with this.

[edwardk]

I too am in Canada (using a 7MB ADSL connection through sympatico) and I have a.5 second delay. I have optimized my stacks, my router settings, I have done the magicfix, I installed the DIVX codec (MJ support asked me to do this) and I still have it

Everything else is great. No noise, static, jitter, etc. Just a slight annoying delay. I use to use Skype Pro before and never had this issue.

[Stewart]

I too am in Canada (using a 7MB ADSL connection through sympatico) and I have a.5 second delay. I have optimized my stacks, my router settings, I have done the magicfix, I installed the DIVX codec (MJ support asked me to do this) and I still have it

In your post here: http://www.phoneservicesupport.com/post11815.html I gave you several suggestions for reducing your delay. The most important one is to not send your calls to Los Angeles and back. Have you tried these suggestions? If so, did you see any improvement?

[SpamBox]

I installed the DIVX codec (MJ support asked me to do this)

Why the heck would one need to install the Divx codec? Does Mj need this? I prefer Xvid over Divx? Will H.264 work give me HD quality phone calls?

[HolmanGT]

I installed the DIVX codec (MJ support asked me to do this)

Why the heck would one need to install the Divx codec? Does Mj need this? I prefer Xvid over Divx? Will H.264 work give me HD quality phone calls?

SpamBox,

I have heard this one before, I have no idea why it would need it. But if it does or would like to have it I game to try (????).

I'll bet MozerD knows the answer to this on but he doesn't come around much anymore. I may just have to email or call him and see if he knows.

[SpamBox]

I have heard this one before, I have no idea why it would need it. But if it does or would like to have it I game to try (????).

When the heck do you sleep? I'm at work (now on the late overnight shift 10pm-8:30am). Was bored today so thought I woudl check out the forum. Been awhile since I've stopped by. Didn't realize it had changed URLs.

[HolmanGT]

I have heard this one before, I have no idea why it would need it. But if it does or would like to have it I game to try (????).

When the heck do you sleep? I'm at work (now on the late overnight shift 10pm-8:30am). Was bored today so thought I woudl check out the forum. Been awhile since I've stopped by. Didn't realize it had changed URLs.

Change URLs and for the better. Now you don't have to wait a day and a half for a screen to come up and your posts and PM don't get trashed half way thru.... Sleep, Hell I don't know my schedule is so screwed up I don't know if it is day or night.

[randie]

As for Canada, we always get screwed. Shaw Cablesystems here in B.C. and Alberta provides their own telephone service under the name Shaw home phone.

It is similar to what some of the big cable companies offer in the USA.

Shaw has their own dedicated network for this service, and freely admits to limiting the ability of their customers to use any voip type service, or the MJ.

I honestly have no idea how they make it that you can surf at blazing speeds, but cannot use bandwith for the MJ, or a Voip service, but again, the clearly admit to doing this.

[JackTheMan18]

As for Canada, we always get screwed. Shaw Cablesystems here in B.C. and Alberta provides their own telephone service under the name Shaw home phone.

It is similar to what some of the big cable companies offer in the USA.

Shaw has their own dedicated network for this service, and freely admits to limiting the ability of their customers to use any voip type service, or the MJ.

I honestly have no idea how they make it that you can surf at blazing speeds, but cannot use bandwith for the MJ, or a Voip service, but again, the clearly admit to doing this.

I am in Toronto, using Rogers Cable as my ISP. Do you know if Rogers is "limiting" VOIP service here as well? I have a slight delay in connecting, and many "dropouts".

[schwim]

Hi there everyone,

I know it was discussed over a year ago, but could one of the techier members tell me if you're applying QoS by application or by port(and if by port, which one?). I'm too new to the system to have experienced any performance problems, but wanted to nix what I could before it happened.

thanks,
json

[HolmanGT]

Hi there everyone,

I know it was discussed over a year ago, but could one of the techier members tell me if you're applying QoS by application or by port(and if by port, which one?). I'm too new to the system to have experienced any performance problems, but wanted to nix what I could before it happened.

thanks,
json

Json,

I have never tweaked my QoS and all has been fine. I personally believe it has more to do with the MJ server you normally default to. And I have seen a few post here that by change their default server most of the audio problems would disappear for the individual making the post.

A problem with using port configuration with MJ is that it uses some ports in the 10,000 to 30,000 range at random. Some have actually opened up all those ports to cater to MJ, personally I don't think that is wise but what the hell do I know. By the way if you don't open up the entire range (and the range I mentioned may not be 100% accurate but close) and MJ attempts to use one you didn't open you problem will return and you'll be back scratching your head.

Also remember that QoS is only a one way street for outgoing data so it will not be much help for incoming.

My personal bullet proof recommendation is don't use you Internet for anything while make a phone call. follow that rule and you won't have any problems at least not any that are created on your end of the pipe.

If you have a large pipe... say 1 up 10 down as I do most all your MJ communications will be great. But remember MJ has it moods (OK so it has a lot of moods) so don't go dismantling your computer every time it acts up a little. Most MJ problems go away on their own if you give it a little time.

Just remember for $20 dollars a year you are going to get lost and lost of free long distance phone call that will be great quality but for $20 a year you may not have 365 days of perfection albeit close.

PS - "The Gospel According to HolmanGT" Never, Never, Never let MJ Tech support talk you into changing or removing software on your personal computer. That is there standard Modes Operandi and usually finishes with your computer broken, MJ not working and the Tech say well I am going to have to escalate this up to engineering and you will get a call in a few days - "I have some swamp land if you are interested". If you have problems come here and follow the instructions of the more seasoned members. It is not as satisfying as talking straight to the horses mouth but the answers you get will be more accurate and I guarantee less painful.

[schwim]

Hi there George and thanks very much for your reply.

I understand your point and it makes complete sense. Luckily we've got substantial available throughput on our end. We've used it a few times so far and the quality has been good. One dropped call and a couple odd artifacts, but as you stated, I've told my wife that there will be some form of trade off for saving over 800 bucks a year. I told her when she gets frustrated at the phone, ask herself if it's worth $800 annually to get rid of the quirk

Thanks again for your response. I imagine you're right that I'll be more likely to run into throughput issues outside of my local network. You know how it is when you get something new though. You try to find all possible weak points. Often it goes a little overboard

thanks,
json

[bern09]

I am from Canada and had this same problem until I talked online with majicjack and they helped me add ports to my firewall so that majicjack could work through or around the firewall. That solved the problem.

[daniel devereaux]

If you "forward" ports 5060 and 5070 for UDP packets on your window's firewall *and* your home router (not Cable/Fios/DSL modem) - most of your voice quality issues should go away!