Caller Can't hear me

Posts: 5 · New Forum Member Joined: Jun 22, 2004 Posts: 5

I've been on Vonage for a year now with no problems until the last week. The people I am talking to cannot hear me - I can hear them, they can't hear me.

HELP! I went to the website Vonage told me to go to & my upload speed was OK.

I have time warner cable and this set up....

>>> Tosihba pcx26000 Cable Moden >>> NetgearMR814 v2 Router >>> Vonage Motorola vt1005v

Posts: 3198 · Posted: Sun May 29, 2005 1:31 pm Post subject:

any recent changes to your setup? any pattern (happens on some calls but not others)?
what are your detailed results from testyourvoip.com boston server and from http://www.vonage-forum.com/voip-speed-test.html

Posts: 5 · New Forum Member Joined: Jun 22, 2004 Posts: 5

Nothing has changed on my system.

My Results:
Download 4,758,224 bps
Upload 367,352 bps
QOS 92%
RTT 54 ms
MaxPause 203 ms

Testyourvoip to Boston:
From Me to Boston
Media Quality MOS 4.2 / 5.0
(Best with G.711 is 4.4)

Degradation Sources
Codec 0.57 76.7%
Latency 0.00 0.0%
Packet Discards 0.17 23.3%
Packet Loss 0.00 0.0%

Codec G.711 (PCM at 64kbps,
20ms RTP payload,
80kbps IP BW)
Round-Trip
Latency 145 ms
Packet Discards 1.1%
Packet Loss 0.0%
Loss Periods Min: 20 ms
Avg: 40 ms
Max: 80 ms
Burst Loss

Jitter Min: 0 ms
Avg: 6 ms
Max: 80 ms

Signaling Quality Post-Dial Delay 78 ms
Call Setup Time 78 ms
Media Delay 156 ms

From Boston to Me:
Media Quality MOS 4.4 / 5.0
(Best with G.711 is 4.4)

Degradation Sources
Codec 0.57 96.8%
Latency 0.00 0.0%
Packet Discards 0.02 3.2%
Packet Loss 0.00 0.0%

Codec G.711 (PCM at 64kbps,
20ms RTP payload,
80kbps IP BW)
Round-Trip
Latency 145 ms
Packet Discards 0.1%
Packet Loss 0.0%
Loss Periods Min: 20 ms
Avg: 20 ms
Max: 20 ms
Random Loss

Jitter Min: 4 ms
Avg: 6 ms
Max: 27 ms

Signaling Quality Post-Pickup Delay 91 ms
Call Setup Time 98 ms
Media Delay 123 ms

Posts: 3198 · Posted: Sun May 29, 2005 1:58 pm Post subject:

hmm ok, those results look more like "I can hear you but you're choppy/breaking up" than absolutely can not hear anything.

Posts: 5 · New Forum Member Joined: Jun 22, 2004 Posts: 5

Usually they can hear me at first, after 20 seconds or so they usually cannot hear me for 30-40 seconds, then maybe they hear me again for a few seconds, then another 30-40 seconds of not being able to hear me.

I don't know much about all this - do you think the problem could be in my cable connection & maybe I need to call them out to look at my service?

Posts: 3198 · Posted: Mon May 30, 2005 8:20 am Post subject:

before I did that I would run pingplotter.com for a while. search if you need more info on it.

Posts: 38 · Vonage Forum Junior Joined: Dec 20, 2004 Posts: 38

Another easy way I found my problem was to open a dos window, then enter the coomand:

ping -t www.google.com

That showed I was having a 12-50% packet loss. The symptoms you describe are similar to the symptoms I had. The problem was with the ISP. In my case it was Bright House.

There is also another problem people seem to be having, where they can hear the person on the other end but the person on the other end can't hear them. Here's a link to one of the threads:

http://www.vonage-forum.com/ftopic6161.html

Posts: 66 · Full Forum Member Joined: Feb 07, 2005 Posts: 66

I am having the very same issues going on over here.

Just Pinging, I get 0 loss... So that is good..

My Stats are also very very good..

Almost makes me wonder if they did an update to my Motorola unit.

ALSO.. This problem is an all day issue. I have also been having lots and lots of lost calls during conversation... (In other words, I doubt it is a bandwidth issue. Could be other issues, but not bandwidth. Not enough people in my area on the cable)

Posts: 66 · Full Forum Member Joined: Feb 07, 2005 Posts: 66

Also, this is the Golden Phone Results. (Actual phone call calculations)

http://www.testyourvoip.com

Call Time 06/01/05 11:58 PM EDT
Origin 832626XXXX
Destination Boston

Audio MOS Details MOS 4.4 / 5.0
(Best with G.711 is 4.4)

Degradation Sources
Codec 0.60 100.0%
Latency 0.00 0.0%
Packet Discards 0.00 0.0%
Packet Loss 0.00 0.0%

Codec G.711 (PCM at 64kbps,
20ms RTP payload,
80kbps IP BW)
Round-Trip
Latency 16 ms
Packet Discards 0.0%
Packet Loss 0.0%
Loss Periods Min: 0 ms
Avg: 0 ms
Max: 0 ms
No Loss

Jitter Min: 0 ms
Avg: 0 ms
Max: 0 m

ALSO Copyed for those who wanna know!

What are latency, jitter, and packet loss and how do they affect my calls?
Once a call has successfully been setup, latency, jitter, and packet loss effects are important predictors of overall call quality.

Latency
A measure of the delay in a call. We measure both the round-trip delay between when information leaves point A and when a response is returned from point B, and the one-way delay between when something was spoken and when it was heard. The largest contributor to latency is caused by network transmission delay. Round-trip latency affects dynamics of conversation and is used in our MOS calculations. One-way latency is used for diagnosing network problems.

With round trip latencies above 300 msec or so, users may experience annoying talk-over effects.

Jitter
Jitter refers to how variable latency is in a network. High jitter, greater than approximately 50 msec, can result in both increased latency and packet loss. Let's see how.

When talking to someone it's important that they hear what you say in the same order that you say it, otherwise they won't understand what you're telling them. Unfortunately, jitter causes packets to arrive at their destination with different timing and possibly in a different order than they were sent (spoken), with some arriving faster and some slower than they should.

To correct the effects of jitter, Voip endpoints collect packets in a buffer and put them back together in the proper timing and order before the receiver hears them. This works, but it's a balancing act. Processing that buffer adds delay to the call, so the bigger the buffer, the longer the delay. Remember the effects of latency? Keep in mind, no matter how big the buffer is, it is finite in size. If voice packets arrive when the buffer is full then packets are dropped and the receiver will never hear them. These are called discarded packets.

Packet Loss
Just as it's important to hear what someone says in the order they say it, it's also important to hear all of what they're saying. If you miss one out of every 10 words or 10 words all at once, chances are you're not going to understand much of the conversation. This is packet loss — some of the voice packets are dropped by network routers or switches that become congested (lost packets), or discarded by the jitter buffer (discarded packets).

Knowing the average packet loss for a call gives you an overall sense for the quality of the call. A call with less than 1 percent average packet loss will always sound better than a call with 10 percent loss. But average loss doesn't tell the whole story. You need to know what type of packet loss you encountered.

There are two kinds of packet loss: "random" and "bursty". Think about two calls each with average 1 percent packet loss. Call A loses one in every 100 packets over the entire call (random loss) while Call B loses 100 packets in two clumps at the beginning and the end of the call (bursty loss). Which call would you rather have? That's why we report not just the average packet loss but also the type of loss and information on any bursts of packet loss during your call (reported as loss periods). It matters.

Posts: 5 · New Forum Member Joined: Jun 22, 2004 Posts: 5

I'm still having the same problems with Vonage. The cable company is supposed to come out today and check my line (of course they were supposed to come out last week and never did). I don't think it is a Vonage problem because I have problems with online games now too.

Man I hope this gets fixed today - I'm worried the cable company won't fix the problem and I'll get ticked off and switch to DSL (even though SBC = Satan Twisted Evil

)