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For wipe call
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of the offline, in
gengral , it
usually apply to
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What is the main
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which one is the
virtual number?
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Trafford Posted:
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diazou Posted:
Hello, It's
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? Thanks!
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Hi all We have
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Av8rix Posted:
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Im trying to add
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Trowski
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PostPosted: Tue Jun 28, 2005 8:44 am    Post subject: Reply with quote Back to top

Pauls point was 100% correct. Tech_girl made a general statement, and it would have been nice to see some information to verify. There is NOT a consistent pattern with one ISP being worse with, and if there was, maybe the company she works for would be that one..that being said...

Denver, try to verify if it a wireless or Vonage issue...I know mine was a wireless issue... old uniden phone plus new router technology equaled a pain in the rear until I bought a new cordless. Please let us know what the outcome was.

Tech_girl, again who is your employer?

And Taylor...someone taught you well...but you would be wise to heed their advice and not say anything at all because your post added absolutely nothing of value to help Denver out, which is why we are here.
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stan_ipkis
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PostPosted: Wed Jun 29, 2005 10:47 pm    Post subject: Reply with quote Back to top

Would someone explain how/why latency and the whole reason of dropped packets are the justification/excuse for choppy and dropped calls when I really think it is because of lousy software/firmware?

Is anyone at Vonage and/or the modem provider looking at writing reasonably good software to make this more reliable? Twisted Evil Mad
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paul248
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PostPosted: Wed Jun 29, 2005 11:03 pm    Post subject: Reply with quote Back to top

I can't comment on the quality of their software specifically, but any Voip system will be vulnerable to packet loss.

With Voip, you have to chop up voice into little packets, and send them to their destination as quickly as possible. If a packet gets lost or delayed on the way, all the software can do is try to cover it up. There's no time to try and retransmit the packet, and any buffering or forward error correction makes the voice take longer to transmit.

Contrast this with, say, downloading a file over TCP. The data is buffered at both ends, and lost packets are retransmitted as necessary, since getting the data there intact is much more important than getting it there on time.

So anyway, if your connection is lagging and dropping packets during a real-time conversation, the only thing you can really do is put up with dropped and lagged voice.

That said, Vonage's software does seem to have a lot of issues, and probably could use some work...
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stan_ipkis
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PostPosted: Wed Jun 29, 2005 11:13 pm    Post subject: Not really Reply with quote Back to top

Can you tell me why a retransmit to reassemble packets in-order is so expensive, i.e. the math please? What is the packet size that these guys chop it up into? Is there a heuristic/any intelligence that looks at the metrics about what packet sizes work better than others for a given device/network?

Also, considering it's udp, not tcp, it has less overhead and I think reassembly with latencies in the 90mS to 150mS will at most take a second. I also think most people can deal with a pause for even a half a second to a second over chops and drops.

My $0,02 and I think it's time one of us started looking into this more seriously.

Sorry, I should have read your reply completely before I wrote this. The part about you not knowing exactly what these guys are doing didn't register. Lol
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paul248
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PostPosted: Wed Jun 29, 2005 11:24 pm    Post subject: Reply with quote Back to top

Well, I guess their reasoning is that latency on voice calls is bad, and they want to minimize it at all costs. If you want a protocol that can handle retransmissions, you automatically triple the latency.

You have to (1) detect that the packet is late, then (2) send a retransmit request back to the sender, then (3) do the retransmission.

Meanwhile, the next voice packets are still arriving, and they can't be played back until after the retransmitted packet arrives, because they must be played in order. If the stream ever stops getting lossy, you still have the problem that the receiver is running a few packets behind, so the receiver either has to drop some packets, or play them back at an accelerated rate.

This is really all in place to avoid latency. Real PSTN phone calls have very little latency, and people have come to expect that for domestic calls.
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stan_ipkis
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PostPosted: Wed Jun 29, 2005 11:37 pm    Post subject: Hmm, perhaps the devil really is in the details :lol: Reply with quote Back to top

Well, with a little more surfing, here's an interesting link that explains a few more things:

http://www.gaoresearch.com/resources/whitepapers/other/voip.php

There are ways to handle packet loss and latencies upto a point. It does take firmware, hardware, good design, and good software codecs.

Makes me wonder if I should be getting off of Vonage and switch to a Voip provider with a better handle on things, or get back to regular old copper until one of these guys get their act together.
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Trowski
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PostPosted: Thu Jun 30, 2005 6:08 am    Post subject: Reply with quote Back to top

You also have to take into consideration the path the call is taking...one never knows what it will encounter along the way...I used to have choppy calls, but then Adelphia and Vonage seem to get things fixed...
It also depends on the codec, your current load in the area if you have cable etc...thats why I go for the upper tier on upload speed..usually around 800 kbps, and have been able to keep on top of any issues adelphia is having.

What type of connection do you have Stan?
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stan_ipkis
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PostPosted: Thu Jun 30, 2005 10:52 am    Post subject: Comcast cable modem Reply with quote Back to top

4mbps up, 384kbps down, and latencies of about 80-90mS to wherever.

Edit: Um, it's switched: 4Mbps down, and 384 up! Obviously!...comcast!


Last edited by stan_ipkis on Thu Jun 30, 2005 3:06 pm; edited 1 time in total
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paul248
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PostPosted: Thu Jun 30, 2005 11:38 am    Post subject: Reply with quote Back to top

Correct me if I'm wrong, but I doubt there's any connection between loss/jitter, and speed caps. A cable modem speed cap only comes into effect if you're trying to send bulk data. An individual packet always gets sent at the maximum rate (10-30Mbps in a cable system), but a cap will delay packets if you're sending too many at once.

I wouldn't be surprised if the performance of Voip were the same between a 128kbit cap and a 1Mbit cap, assuming that nothing else is using the connection.
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Trowski
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PostPosted: Thu Jun 30, 2005 11:42 am    Post subject: Reply with quote Back to top

My understanding that the upload is the factor not the DL...if you had 128 kbps and had anything running, then yeah that could be an issue...Most places I have read that 128 should really be a minimum for VOIP...I could be wrong, so maybe Dan or someone could help clarify...

But for me, bigger is always better! Wink I run two lines, an online DB, and multiple windows so I need a good UL speed.
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