Considering that I was able to get an RT31P2 router with the bandwidth set to 30Kbps to work over a dial-up connection, I highly doubt upgrading to 4MB is necessary. What's your upstream rate?
I will note, the RTP300's don't currently use the bandwidth saver, just a full 90Kbps all the time. Awesome fax & data connections over it though.
The up stream currently is 250kb/sec, the up stream is good, its the latency, and high packet loss issue, that we have to deal with. All of this started August 1st 2005. We can use the phone between the hours of 1:30 AM to about 5:30 - 6:00 AM. Between these hours the phone is clear as a bell, with no chop. After that the chop starts out dropping letters in words and progresses to the loss of complete sentences. I can not say this enough, the Vonage techs have been wonderfull! Thank you guys and gals.
It is the same for me. I can use the phone until about 7:00 AM or so.. clear.. and then during the conversation.. I have problems hearing the other person.. every 4 seconds or so.. there is a gap in conversation... The person on the other end has no problems hearing me.. So it appears that voice coming in has problems.. This appears to coincide with test results from testyourvoip.com MOS Analysis from You TO Boston 4.2 / 5.0 (Best with G.711 is 4.4)
These results were generated at 08/07/05 05:47 PM EDT
Quality Between You and San Jose Your score: 1.0
Forget the phone, try Pony Express | Like tin cans and string 1.0| As bad as a crummy cell phone call 2.0| As good as a decent cell phone call 3.0| Like calling next door 4.0| Better than being there! 5.0
Test Details The information below explains why your call quality score (MOS) was less than perfect.
MOS Analysis from You TO San Jose
Media Quality MOS 4.2 / 5.0 (Best with G.711 is 4.4)
Codec G.711 (PCM at 64kbps, 20ms RTP payload, 80kbps IP BW) Round-Trip Latency 119 ms Packet Discards 1.2% Packet Loss 0.0% Loss Periods Min: 20 ms Avg: 20 ms Max: 60 ms Random Loss
Jitter Min: 0 ms Avg: 6 ms Max: 49 ms
Signaling Quality Post-Dial Delay 63 ms Call Setup Time 63 ms Media Delay 156 ms MOS Analysis FROM San Jose To You
Media Quality MOS 1.0 / 5.0 (Best with G.711 is 4.4)
Codec G.711 (PCM at 64kbps, 20ms RTP payload, 80kbps IP BW) Round-Trip Latency 119 ms Packet Discards 0.0% Packet Loss 36.9% Loss Periods Min: 20 ms Avg: 280 ms Max: 960 ms Burst Loss
Jitter Min: 3 ms Avg: 7 ms Max: 20 ms
Signaling Quality Post-Pickup Delay 70 ms Call Setup Time 79 ms Media Delay 101 ms
Whoa... a 1.0!!! I'm getting ~3.4-~3.6 over the dial-up connection. Something's definitely screwed up there. All packet loss, I see, but not 100% packet loss... so maybe the ISPs just waaay oversold compared to the bandwidth they have.
But why did our phones work untill August 1st? Here is something worth reading from the FCC dated 08/05/2005, FCC Adopts Policy Statement, New Principles Preserve and Promote the Open and Iterconnected Nature of Public Internet. http://hraunfoss.fcc.gov/edocs_public/attachmatch/DOC-260435A1.pdf I thought I would share. When I run PingPlotter on my end it does show 100% packet loss.
It's 2:58 PM and the phone is still working. Received a call back from city hall cable liaisons office. We explained the situation, and all the problems with the cable company, including how rude the local office personel were, and the song and dance routine from the supervisors over the phone. Good luck to those of you still struggling with their ISP providers. We are the ISP's customers, and we have rights!!
That was the reason I started this thread, was to see if anyone else had the same issue with the same ISP. What really set me off was the way the cable company passed it off as not their problem (we don't support that), and you should get our service.
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