Back Again With Voice Breaking Up at the Other End..I give up

[VillageGuy]

From the start I have had issues with calls breaking up at the other end (meaning my voice).
Have had numerous support deals with VOIPO. They have gotten on line with my router and made changes putting ATA in the DMZ etc. It appeared it was working but for short period of time.
Bought the OBi110 ATA for Google Voice ended up switching it out with the Grandstream.
Was told if the output on the Grandstream was blown on the house wiring I would be charged for another unit.
At one point had both ATA units working off my router. Then the Grandstream wouldn't ring the phones but had dial out tone. Voipo figured out that the two units couldn't exist on the same router.
NOW I have the Grandstream replaced with the OBi110 and still calls are breaking up on the other end. VOIPO will not support the OBI110 either.
Ridiculous that the Grandstream or VOIPO won't allow connection to the house wiring. The OBi110 will handle house wiring.
Have checked internet speeds and they are fast for up/down. Router is connected to comcast cable.

I have resolved that the problem with breaking up will never be resolved. Sick of numerous attempts with VOIPO technicians. Have to say supervisors were as helpful as they could be.

Won't be renewing my contract with VOIPO again. I had voip with T-Mobile for many years absolutely NO problems. Sorry I switched to VOIPO.

[voipinit]

One thing you might want to try is putting your ATA ahead of the router for testing purposes. If the breaking up goes away, then it is something in your router. I am not familiar with the OBI110 but I would suspect they are not up to Grandstream capabilities. Any weakness in you system will appear with VOIPo as they don't compress the audio stream (this is a good thing as it gives the best audio quality). You might also ask VOIPo to change you to a different CODEC to one that uses compression (again for testing), there will be a slight degradation in audio quality but it is slight and most people can't detect the difference. Just throwing some ideas out there, I know it can be frustrating. Let us no how it goes.

[tritch]

Good suggestions given "voipinit"..........

Problems with choppy voice can almost always be attributed to network congestion. In your case it sounds like upstream congestion on your end, since the other caller hears the chop. If the the upstream congestion is within your own network, you may want to consider setting up QOS within your router or alleviate any other Internet activity during your call. It's most likely congestion outside of your control (high latency, high jitter, bandwidth saturation, etc.) with your ISP (Comcast) especially during the peak hours of 4pm-11pm. The best way to test this is to take your ATA over to one of your family/friend's house who are on another ISP and hook it up there. Test it again and see what happens. Voipo has no control over your ISP congestion problems......

BTW, my dad also has Voipo and experiences choppy calls during 4pm-11pm. Like clockwork, every night his upstream connection goes downhill. His cable ISP recently oversold his area about 6 months ago and just about every night the chop starts up. He doesn't hear any chop, but the other callers do. He can take his ATA over to my house and the chop is completely gone.....

[VillageGuy]

tritch-Agree that the problem most likely upstream congestion. The Grandstream and the OBi110 were setup in the DMZ on the router there for by passing the firewall etc. VOIPO set it up. I was having problems today in the morning. Interesting though, I had T-Mobile voip for years and no issues.Could be the router not sure what more I can do. QoS is also setup giving highest priority to the LAN port the ATA is attached too. Problem is intermittent.

[pcri2013]

Hi VillageGuy,

I too have nothing but problems ever since I switched from Verizon to VOIPO. I tried all of the settings suggested by the VOIPO tech support personnel, but the garbled sound, dropped calls, and echos continue to persist. I even upgraded my FIOS internet service to 50/25 mbps and my speed test shows that I am getting 60/30 mbps. VOIPO keeps saying that it is the latency of my internet service, but I never had any slow response from any of my internet devices.

I am really getting frustrated with VOIPO's inability to solve this problem so much so that I may have to find other providers. I would hate to go back to Verizon, but I may have to do that.

Sorry for venting here... I usually don't post in any forum, but I wanted to share my experience with my fellow VOIPO subscribers. Good luck to you, VillageGuy. I will find out more about the T-Mobile VOIP service since you had good experience with them.

[JacobsLive]

Hello VillageGuy & PMpcri2013,

Sorry to hear that you guys having issues with your service. With verizon FIOS chances are nearly nill for latency issues. Only possibility is a faulty modem. I had verizon FIOS in the past with 35/35 connection and the Actiontec router had to be replaced at least in every 3-4 months. If you have latency with FIOS service, it should be your modem.

I would suggest both you guys do a VoIP test on your lines especially as soon as you notice the problem. Here is the best place to do a VOIP test:
http://myspeed.visualware.com/indexvoip.php - select North America from the map and select appropriate server closer to your location.

If you have a jitter rate above 50 ms and packet lose anything above 0.0%, there there is a problem and your connection is not good for VoIP service. If the result shows otherwise, contact support again with the test results and they should be able to fix it.

I have seen echo and poor quality of incoming voice with my service as well for the last couple of days. As of now it happens only occasionally and I am still observing the situation and trying to isolate whether it is VOIPO or any other factors. Good luck to you guys.

Here is an example of my test results:
Test Type: Voice over IP
Downstream Jitter: 0.13 ms
Downstream Loss: 0 %
Downstream Packet Order: 100 %
Upstream Jitter: 1.26 ms
Upstream Loss: 0 %
Upstream Packet Order: 100 %
Packet Discards: 0 %
MOS Score: 4.2

[pcri2013]

Hi JacobsLive,

Thank you for providing the link to the VOIP test.

I ran the test and here are the results:

VoIP test statistics
--------------------
Jitter: you --> server: 0.0 ms
Jitter: server --> you: off
Packet loss: you --> server: 0.0 %
Packet loss: server --> you: off
Packet discards: 0.0 %
Packets out of order: 0.0 %
Estimated MOS score: 4.2

Since jitter is 0.0 ms and packet loss is 0.0 %, I don't think the problem is with my internet connection. This confirms the speed rating that I have seen using Verizon's website.

[VillageGuy]

Ran the test and everything is good:
Voip Results
Upstream jitter 3.5 ms
downstream jitter 4.8ms
No packets lost upstream/downstream
MOS 4
Register ms 539ms
Invite ms 544ms

[pcri2013]

I ran the test again, this time with the server software installed. The results look good.

Upstream jitter 0.0 ms
Downstream jitter 0.2 ms
Upstream packet loss 0.0 %
Downstream packet loss 0.0 %
Upstream packet order 100.0 %
Downstream packet order 100.0 %
Packet discards 0.0 %
MOS 4.1
REGISTER ms 16 ms
INVITE ms 25 ms
BYE ms 15 ms
Test Voice-over-IP

[chpalmer]

I usually get a B when it comes to those java tests... Yet I have absolutely no problem with any of my Voipo lines. Makes me curious if routers, or actual ISP routing have more to do with it in these cases.

I gave up on home (SOHO) style routers years ago. They're just too unpredictable for my standards...

Good Luck guys!

[tritch]

Makes me curious if routers, or actual ISP routing have more to do with it in these cases.

Exactly..

My general reference to network congestion above can also mean anything that happens to your voice data packets once it traverses the Internet with your particular ISP. Every ISP is different on how they "route" their data around the Internet. They each use different switches, routers, backhaul, etc. For example, I'm on CenturyLink's DSL service. If my data needs to get from Houston to Dallas, it will basically make a few hops through a couple of local switches/routers then uses CenturyLink's own backhaul fiber (or Qwest) to reach Dallas. If I were to use another ISP (like Comcast), they are likely to use different switches/routers and backhaul provider to route the data to Dallas. It may even traverse through several more hops to different cities just to reach Dallas. You're dealing with the Internet and everybody's routing is different from one ISP to another. In addition, the routing will be different from the same ISP from different cities. There's also the occasional outages/congestion problems within their network or backhaul which may cause them to "re-route" the data from one day to the next. There are a lot of variables going on behind the scenes. This is why one can experience completely different results from one voip provider to another simply because of the call "routing" it takes with your particular ISP.

Voice packets need to be in real time, so any type of hiccup or delay in the whole routing process will cause chop/garble.

BTW, those voip tests only give you a "general" idea of how well voip may work with your network. It does not take into account the data "routing" your call may make.

Hopefully, Voipo's support team will try to do everything possible to keep you as a customer. Unfortunately, there are a few people that have network issues that are outside their control and they simply can't resolve it no matter what they do......best of luck!

[JacobsLive]

Hi pcri2013, I saw your both test results and I think there is something is not right with your connection/Java installation/PC/Router - It is hard to predict what is at fault. If you notice, your first test did not connect to the jitter server, but your second test did. I have a reason to believe it is your router because of your voice quality fails at times. I could be wrong, but just my personal opinion.

VillageGuy, your connection seems to be having too much buffering, It took more than 500ms in your test to get a handshake from servers, which could be affecting your call quality as well.



BTW, those voip tests only give you a "general" idea of how well voip may work with your network. It does not take into account the data "routing" your call may make.


I fully agree, to find out exact call quality and conenction issues VOIPO support can initiate a call monitoring on your account and they might be able to shed more light into finding a right solution, if not already.

Hope things will work fine with you guys soon!