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nixie21
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PostPosted: Sun Nov 14, 2004 9:07 pm    Post subject: RT31P2 QOS? Reply with quote Back to top

I left the default settings. What does everyone use for their qos settings???

thanks
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quisp65
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PostPosted: Mon Nov 15, 2004 8:29 am    Post subject: Done just a little experimenting Reply with quote Back to top

I've done just a little experimenting when I saw your message because I would like to know more about QOS with the RT31P2. I got Charter 3mb/256K and when downloading full speed from newsgroups I would notice the telephone conversation would cut in and out. This would do it if I used the default of "manual 256" or switched it to "auto detect" or increased it above my upload speed to "manual 384". I wonder if the QOS even works?!! My download speed of my newsgroups didn't change in any of the calls. I am using the middle codec on the bandwidth saver and usually get perfect telephone service except for the occasional one side can't hear the other one at all call. Could try the g729 codec but I don't download much so this isn't a huge issue.
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reebok
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PostPosted: Mon Nov 15, 2004 9:09 am    Post subject: Reply with quote Back to top

I tested an ftp upload. I was getting around 45K/s. I tried with Packet8 (since I have them still, waiting for lnp) and there was no QoS involved. It still did fine. When I tried with automatic on the RT31P2, it throttled my ftp upload speed to about 15K/s. I don't remember if I did it with manual, but I assume it would throttle your uploads to 256kbps or whatever you set it to. I had port and application QoS disabled. Voice QoS was the only one enabled.

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quisp65
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PostPosted: Mon Nov 15, 2004 9:56 am    Post subject: Reply with quote Back to top

Does having the "QOS upstream rate" to "manual 256" suppose to set aside 256K of upload bandwidth for telephone conversation (like I thought) or does it only allow 256K upload for the other applications? The manual doesn't discuss this issue.
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scubaduba
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PostPosted: Wed Nov 17, 2004 8:05 pm    Post subject: Reply with quote Back to top

Good question. I'd like to know more about this too.
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TheMole
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PostPosted: Wed Nov 24, 2004 11:06 pm    Post subject: Reply with quote Back to top

Reebok? Sort of off topic question. Why are you leaving Packet8? I heard good things about them and was just curious.

Anyways the QoS of the RT31P2 works correctly. Do the following:

1.)Keep your codec set to 90kbps
2.)Change the Voice QoS on the RT31P2 to Manual and at 180kbps (this will allow for a 3 way call or 2 calls on both call legs simul. without voice degradation due to bandwidth

3.)save the settings and talk on the phone. while talking run a speedtest and you will notice your advertised upload speed will show less the QoS (180) from the rt31p2

**this is only when the rt31p2 is directly connected downstream of the ISP CPE**

Mole

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reebok
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PostPosted: Thu Nov 25, 2004 9:28 am    Post subject: Reply with quote Back to top

Packet8 is ok, but their volume is way too loud for me, I like caller I'd with name, it's a little less stable than Vonage, and I like online voicemail. Most of these complaints are nore personal in nature than an overall dislike of their system. Also, I'm not paying for it. If I was, I would likely stay with the cheaper Packet8.

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paul248
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PostPosted: Sat Nov 27, 2004 4:00 am    Post subject: Reply with quote Back to top

quisp65 wrote:
Does having the "QOS upstream rate" to "manual 256" suppose to set aside 256K of upload bandwidth for telephone conversation (like I thought) or does it only allow 256K upload for the other applications? The manual doesn't discuss this issue.


Note: everything below this line is speculation, but I think it should be mostly true.

The number you type in is the amount of bandwidth that you have available. So, if you've got 256k upload, then you should type 256 in the box.

When you're on the phone, it basically limits your total upload speed to 256k so that nothing gets queued up by the cable modem's capping system (assuming you have cable). Then, all the voice packets are given top priority by the router. When you're in the 90k bandwidth mode, then this has the effect of allocating 256-90 = 166k for everything else.

If you set the value to something really low (10k for example), then it will assume you have no extra bandwidth available beyond what the phone requires, and effectively shuts off the internet connection for the computers while you're on the phone. When I set mine to 10k, I couldn't even get pings to go through, but the voice was just fine.

When I disabled voice QoS completely, and attempted an FTP upload, the outgoing voice data got completely mutilated, as I expected. The voice was very choppy and robot-like, and was lagged by about 1-2 seconds. However, the incoming voice sounded normal.

QoS set to Auto seemed to behave exactly the same as 256; In both cases, my FTP uploads dropped from 32kB/sec to 9kB/sec while I was on the phone, and the quality of the phone call was unaffected.
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paul248
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PostPosted: Sat Nov 27, 2004 4:04 am    Post subject: Reply with quote Back to top

I should also add that the Voice QoS setting seems to have absolutely no effect whatsoever when you're not making a phone call.

In the end, I decided to just set it to Auto.
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PostPosted: Fri Dec 03, 2004 5:21 pm    Post subject: Reply with quote Back to top

TheMole wrote:
2.)Change the Voice QoS on the RT31P2 to Manual and at 180kbps (this will allow for a 3 way call or 2 calls on both call legs simul. without voice degradation due to bandwidth
I spoke with a Vonage tech rep, who admitted, he didn't know a great deal about telco stuff (which is why I had called in the first place regarding FSK VMWI), but he knew a great deal about VoIP/SIP routing.

I basically asked this question, quoted above. The Vonage rep stated that you will not need any more than your set upstream bandwidth as set in the dashboard. He said that even in the case of call-waiting and 2 incoming calls, the 2 incoming calls will use the downstream bandwidth (say to 180Kb/s), but your up will still be the same, since you are still only talking with 1 line at any time. He did say there may be some momentary latency lag when switching back and forth between callers, but most users would never notice.
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