VoIP User sound quality feedback

Quote:

Any tips?

Hi Mark,

Most likely a bandwidth issue (our gateway is on a level 3 backbone, so when you call via the gateway. at least one half of the connection will be fast).

When you call peer to peer, try to ensure QoS by not having any other web applications open at the time (downloads, email etc).

Dean

Thanks Dean

Any other tips settings codecs etc?

Mark

No problem with the sound, but sometimes I feel like the sound is not full duplex, I'm using a headset with microphone.... But the sound is excellent expect this. Thanks for a great service

Hi i'm a new user from italy,
Today for the first time i tried 3 phone calls throught your service to 3 friends here in italy.Total time:about 25minutes.
The sound quality is really good!!!Great work, i wasn't able to find any difference with my pstn line

Regards
Skumpic

We are using the outbound service quite regularly now particularly for short calls (avoids paying BT 5p for a 10 second call). Working hard to keep the inbound usage up as well as the PSTN -> sip usage. I have noticed that in the last week or 2 that the sound quality is getting worse, stuttering and having to say pardon alot. Is this due to more users (I keep recommending friends) or is this more likely to be general internet traffic. I am using an ATA and POT and avoid surfing etc during calls.

Keep up the good work

Dave

It's not likely to be at our end Dave, we haven't even scratched the surface of total capacity yet, and when/if we do, we'll just bolt in another Cisco.

I'll look into the "Donald Duck" comment you made on the dial-in dial-out numbers though - that sounds codec related to me. Certainly shouldn't do that PSTN to PSTN. Thanks for the heads-up.

Dean

Great Job!! Sound quality is very great better than skype on my side.

thank you so much for this great service.

Hi

I am a new user from Asia/Malaysia . I tried making a few calls yesterday to US numbers , the call quality is perfect , no obvious difference to landline.

But calling to a PSTN number in Malaysia would generate a echo that I can only hear (i am hearing my own voice echo back to me) , the receiver of the call did not hear this and is clear from his end.

Also there is a 0.5 - 1 second delay in the conversation for calls to PSTN malaysia , to US , this is not noticeable and the call is perfect .

Hi csleong and welcome to the forum.

Thanks for taking the time to give us some feedback.

The delay you experience is caused by a delay in the routing between our PSTN gateway (in London) and the peer PSTN gateway in Malaysia. Routing between the two is over the internet.

You may find if you call this evening, there is no delay. Or you may find that it's worse - it all depends on how well the internet routing is between London and Malaysia at the time.

My experience is that some parts of the globe (especially South East Asia) are less well equipped in terms of internet backbone than others. It doesn't surprise me that you didn't experience any delay calling the US for example. North America has the largest internet backbone structure on the planet. Consequently it's very fast.

Quality of internet routing in any given continent will have a dramatic effect on the quality of VoIP service attainable.

Dean

Since I have started using VOIP I have found that I get very variable voice quality.

My setup is BT half meg ADSL with a BT Wireless Network 1250. To this I connected an XP Home Addition PC and I run X-LITE.

Initially I connected to the Router using USB 802.11b even though the Router is located in the same room as I have 2 PCs and only one Ethernet Port on the Router. With this setup I noticed that I was getting packet loss due to the wireless link which showed up as short drops in voice quality and XP telling me it had lost the LAN connection.

Since then I have tried various things to fix the problem. I changed the WI-Fi channel in case of interference. This made no difference.

I fitted an old 10 Mbps Ethernet Card and ran comparisons between Ethernet and WI-FI by using www.testmyvoip.com. This showed an improved MOS score for Ethernet (as you would expect as WI-FI introduces delay even without my interference problems).

I was also told that the poor performance of my WI-FI could be caused by other devices the use the same frequency eg the radio devices used for sharing TV Satalite Decoders.

As a result of these tests I have purchased an Ethernet Switch and a 100 Mbps lan card and Voice Quality is improved. I am regularly geting MOS scores of 4.2.

So if you are having problems it could be due to Wi-Fi.

Hope this is of help.

Mark

Hi,

The quality of what I say is perfect, yet the incoming quality is awful (barely completeing a full word...) - I have tested it by making a few recordings.

Any ideas?

Also, my 0844 number provided with my account just gives an engaged tone.... :-/.

Is it possible for the number used for outbound calls to be one of the numbers selected in 'My Numbers' that points to another tel num?

Thanks,
Brad.

SAPnet :

Hi, The quality of what I say is perfect, yet the incoming quality is awful (barely completeing a full word...) - I have tested it by making a few recordings. Any ideas?

You do not say what connection you are using.

Do you share your connection if you do then if another user is downloading a file or streaming video then you may not have the available bandwidth i/c. Suggest you try with only your PC or VOIP Phone using the connection.

Hope this is of help.

Mark

The voice quality is excellent (using EyeBeam) and exceeds even some of the paid services.

I think for whatever reason, SIP to SIP calling is something that needs work.

Please keep it up!

zerogravity :

The voice quality is excellent (using EyeBeam) and exceeds even some of the paid services. I think for whatever reason, SIP to SIP calling is something that needs work.

Hello Zerogravity,

What, according to you, would you think need changing/work in the SIP to SIP calling ? Please explain some more as we are allways looking at ways to improve our services.

Thanks,

Tj

Hi Tj:

The couple of times I tried SIP-to-SIP calling using G729, the voice quality tended to bit unstable and there were patches of less than desirable voice quality. SIP-to-PSTN works like a charm and I guess I wish somehow that you can replicate the same in SIP-to-SIP.

It is still much much better than some other service providers. Even paid services like Terracall which do allow SIP-to-SIP as well fared very poorly. Probably they want us to use SIP-to-PSTN more rather than SIP-to-SIP.

I guess more people can use SIP-to-SIP when cost effective and user friendly (plug and play style) ATA's become available and people can finally talk just the way they are always used to.

Hope it helps.

Thanks,

Kasauli