AGEphone Gadget Beginners Tutorial

You would be surprised how many of our AGEphone Gadget customers ask us how to get started with the program, some having bought the softphone without even giving it a try beforehand. While we appreciate such enthusiasm for our products we urge you to give the trial version a test before spending your money. You'll get all the features of the full version, can test the softphone for as long as you want and the only difference is that each call is limited to one minute. This should be sufficient to check out both the sound quality and whether or not your internet telephony provider is working without problems.

Speaking of which, the single biggest point of failure is the provider configuration. As many of you are starting from scratch here, let me provide you with some details! For starters, the AGEphone Gadget does not work without an internet telephony or so called Voice over Internet Protocol ("VoIP") provider. It is just like your regular phone in that regard: If you don't have a provider it is pretty much useless. You can still make calls in Peer-to-Peer ("P2P") mode to other internet telephony software or devices, but as you probably rather want to place calls to regular landlines and mobiles it is time to start looking for a decent VoIP provider. If you expect yet another contract and high extra costs, rest assured that you are in for a pleasant surprise called

There are tons of internet telephony providers out there, some cheaper, some offering more features, some open, others secretive. This comparison page has proven to be a good starting point for most of our customers. It lists some of the most competitive prepaid price plans that don't require any monthly commitment. The company behind these plans is called Betamax (just like the old Sony video tape standard) and while they don't offer much when it comes to customer service, they undercut your old provider and Skype by a multiple for many destinations. Compared to Skype, most VoIP providers support the open Session Initiation Protocol ("SIP") standard and can be used with any compatible hard- and softphones.

Once you have chosen one of the price plans, please register on their respective page. There is no need to charge up your account just yet because you get a few free test calls each day to the locations marked with a "0". These calls usually last for about a minute and thus are a perfect match with our trial version. In order to use use your provider of choice with the AGEphone Gadget, start up the softphone and go into the settings. You are greeted with the following screen:

...which can be a bit confusing the first time you look at it. Don't worry, this is just because some VoIP providers use very complicated settings to make it difficult or impossible (e.g. many internet service providers that offer their own VoIP services) to use them with 3rd party products. The AGEphone Gadget supports nearly all of these cases, but luckily Betamax has nothing to hide which greatly simplifies the configuration. Let's pick their "voipcheap" plan as an example. Go to their homepage and click on "Instructions" in the top bar. Select "Using a SIP device" from the table of context after that and folllow the "click here" link. The next page brings up detailed 3rd party device and software setup instructions in which the most interesting point for us is the "General" section that lists

SIP port : 5060

Registrar : sip.VoipCheap.com

Proxy server : sip.VoipCheap.com

Outbound proxy server : leave empty

Account name : your VoipCheap username

Password : your VoipCheap password

Display name/number : your VoipCheap username or voipnumber

Stunserver (option) : stun.VoipCheap.com

This may again look pretty complicated the first time you see it, but let's translate that list into the AGEphone Gadget settings and you will see that it is not that bad:

Realm: VoipCheap.com

SIP Domain: VoipCheap.com

SIP Server: sip.VoipCheap.com

Registrar: sip.VoipCheap.com

Register Expires: 3600

Please note that all the above strings are case sensitive and a slip will prevent AGEphone Gadget from connecting to the provider successfully. The "Realm" and "SIP Domain" are usually the "SIP Server" or "Registrar" string without the "sip." part. This is at least valid for all of the Betamax providers and so you can take this information and duplicate it with their other plans.

Once you have filled in the fields, please click the "Account" button and you will be shown the user credentials page. Please enter your Betamax user name into the "User ID", "Auth ID" and "Display Name" fields and also add your "Password":

The configuration is more complex than it needs to be in this easy case, but some of the other non-Betamax providers that may require different settings for these fields.

A click on "Network" brings up a more advanced part of the server configuration which you can safely leave at the defaults shown below. You can enter "stun.VoipCheap.com" into the secondary "Sec. STUN" field as a backup so that the AGEphone Gadget will work without problems even if the primary server fails for some reason.

Click on "Other" after that to see the fourth and last settings page which shows a field to enter the serial number that you are going to need once you want to convert the trial into a full version. It also shows the "Direct Sound" option that might improve the sound quality a little if set. If you are interested in doing so, we recommend checking the box after you have successfully connected to your VoIP provider for the first time.

After that Hit the "OK" button and that is it for the basic configuration. Give it a few seconds and the AGEphone Gadget should show a "Ready", indicating that you can now place a call. Please remember that you need to use the international "+XX" / "00XX" number format for all contacts dialed with Betamax (and 95% of all other VoIP providers) without any exceptions for local numbers in your own country.

I hope this little tutorial helps you to get your internet telephony provider up and running. If after reading this you still have trouble getting things to work please always feel free to leave a message in our the forums, chat with me here, drop me a mail to michael // ageet // com (you know what to make of this) or take a look at our support contact page.

Turn up the Buffers a Notch (or two)

I've had an interesting chat with a customer yesterday about audio quality... or the lack thereof. He was pretty displeased with what AGEphone Mobile 2 had to offer in that department and after hearing a sample that he recorded on his answering machine and sent in I could see his point: His voice was distorted to the point of being unintelligible and so we started going through possible causes -

1. High CPU time consumption: This is the less likely reason as AGEphone Mobile 2 has been found to perform well even on older WM5 devices with 200 MHz CPUs. We also have fixed a bug in the past that could cause CPU spikes on some devices and so the only possible cause is 3rd party software so completely hammering the device that it chokes all other programs. If you suspect any such program on your device, Task Manager is a great program to make sure.

2. Network issues: Here we're talking about a large field that ranges from insufficient connections over network congestion to mobile providers purposely slowing down VoIP traffic. Our mobile softphone needs at least a standard 3G connection that offers 128 kbit/s up- and downstream with G.711 and 64 kbit/s for GSM-FR. While both codecs theoretically need less bandwidth (64 + 13 kbit/s respectively), the packet overhead (plus 21 kbit/s) and a certain security margin need to be accounted for. Thus there is absolutely no chance for you to run AGEphone Mobile 2 over GPRS. And if you say "But I am using EDGE!" now, that one is also less than ideal because even if the bandwidth is sufficient, the latency introduced (around 500 ms) is way too high to have satisfactory calls without constantly cutting each other off.

3. Settings: And here namely the buffer settings. Even if your connection is sufficient, different devices and networks call for different buffer settings. Raising either the incoming or outgoing buffer can fix stuttering and noise in the respective direction.

As it turned out our customer was using a SE Xperia X1 with CPU monitoring over his home WiFi that supplies way more than the required bandwidth. We could thus rule out point 1. and 2. and I advised him to raise his outgoing buffer.

In order to do that he had to open the AGEphone Mobile 2 settings and go to "3. Audio" where we have the "Playback Buffer" (incoming) and "Record Buffer" (outgoing) values. Raising the latter to 100 ms made the sound intelligible and going up further to 180 ms fixed the sound quality issues completely.

Some of you may need different values -- I myself have set the values to 120 and 60 ms respectively which gives me more than adequate quality. Sometimes you may have to raise the incoming or "Playback Buffer" if the sound you hear is skippy or distorted. The "Playback Buffer" goes up in larger steps as the maximum values needed can be considerably higher than for the "Record Buffer".

Setting the buffers is always a tradeoff between latency and audio quality. If you set them too high the delay between you and whomever you are talking to could become too high to have a good conversation. You generally want to keep the delay below 500 ms, a figure that adds up when you combine:

Connection Latency (CL) + Buffer (B) + Distance (D)

If you are using WiFi, connection latency is probably a minor factor, but if you use a mobile broadband provider you can add 100 - 300 ms depending on which mobile broadband technology you use (3G or 3.5G) and how far you are from the next cell tower. The buffer is a combination of the incoming and outgoing value to which you have to add the distance between you and the callee. Let's assume I place a call from Japan to Germany over my HSD(U)PA (3.5G) enabled device:

CL 110 ms + B (120 + 60 ms) + D 210 ms = 500 ms

The call should be reasonably fluent without any walkie-talkie effect. If you don't usually place your calls to the other end of the world and / or if you are using WiFi you should be able to get much better values which leave some more room for higher buffer settings if required. With this knowledge in hand, just experiment a bit for yourself to find the optimum buffer values for your everyday use!

AGEphone Mobile Beginners Tutorial

You would be surprised how many of our AGEphone Mobile customers ask us how to get started with the program, some having bought the softphone without even giving it a try beforehand. While we appreciate such enthusiasm for our products let me advise you to give the trial version a test before making your purchase. You'll get all the features of the full version, can test the softphone for as long as you want and the only difference is that in the demo each call is limited to one minute. This should be sufficient to check out both the sound quality and whether or not your provider is working without problems.

Speaking of which, the single biggest point of failure is the provider configuration. As many of you are starting from scratch here, it's time for some details! For starters, AGEphone Mobile does not work without a internet telephony or so called Voice over Internet Protocol (VoIP) provider. It is just like your regular mobile phone in that regard: Remove the SIM card and it is pretty much useless. You can still make calls in P2P mode to other softphone enabled computers and mobiles, but as many of you probably want to place calls to regular lines it is time to start looking for a VoIP provider. If you expect yet another contract and high extra costs, rest assured that you are in for a nice surprise --

There are tons of internet telephony providers out there, some cheaper, some offering more features, some open and secretive, others open. This comparison page has proven to be a good starting point for most of our customers. It lists some of the most competitive prepaid price plans that don't require any monthly commitment. The company behind these plans is called Betamax (just like the old Sony video tape standard) and while they don't offer much when it comes to customer service they undercut your old provider and Skype by a multiple for many destinations. Compared to Skype, most VoIP providers support the open Session Initiation Protocol (SIP) standard and can be used with any compatible hard- and softphones.

Once you have chosen one of the price plans, please register on their respective page. There is no need to charge up your account just yet because you get a few free test calls each day to the locations marked as such. These calls usually last for about a minute and thus are a perfect match with our trial version. So you can use your provider of choice with AGEphone Mobile start up the softphone, click on "Menu" and then "Settings".

Next go to point "1. SIP Accounts",

click "Options" and then "Add".

Alternatively, if your chosen provider is already in the list you can click "Options" and "Edit". Either option brings you to the detailed provider configuration. On its first page you have to fill in

which is more detailed than in many other softphones. This is because some VoIP providers use very complicated settings in order to make it difficult or impossible to use them with 3rd party products. AGEphone Mobile supports all these cases, but luckily Betamax is not one of those providers which greatly simplifies the configuration. Let's pick their "voipcheap" plan as an example. Go to their homepage and click on "Instructions" in the top bar. Select "Using a SIP device" from the table of context after that and folllow the "click here" link. The next page brings up detailed 3rd party device and software setup instructions in which the most interesting point for us is the "General" section that lists

SIP port : 5060

Registrar : sip.VoipCheap.com

Proxy server : sip.VoipCheap.com

Outbound proxy server : leave empty

Account name : your VoipCheap username

Password : your VoipCheap password

Display name/number : your VoipCheap username or voipnumber

Stunserver (option) : stun.VoipCheap.com

This may look pretty complicated the first time you see it, but let's translate that list into the AGEphone Mobile settings and you will see that it is not that bad:

Please note that all the above strings are case sensitive. The "Realm" and "SIP Domain" are usually the "SIP Server" or "Registrar" string without the "sip." part. This is at least valid for all of the Betamax providers and so you can take this information and duplicate it with their other plans.

Once you have filled in the fields, please click "Next" and fill in your user credentials on the page that shows up. Please enter your Betamax user name into the "User Name", "Authorization" and "Display Name" fields and also add your "Password".

The configuration is again more complex than it needs to be in this easy case, but there are other providers that may require different settings for these fields.

A click on "Next" brings up the detailed part of the server configuration which in our case is safe to be left alone. If you are still interested in what all the options mean, a look into the settings manual can probably answer your questions. But for now simply click "Next" and then "Finish" and you are back in the provider selection dialog. Since most Betamax providers don't offer a dial in number for other people to call you there is no need to check the box left of your newly created provider. Should you have a dial in number with your provider, please make sure to check this box.

Click on "OK" now to return to the main menu and go into the "2. Network" options.

Enter "stun.VoipCheap.com" into the "Primary STUN Server" field. Should that field be greyed out, please make sure that "STUN / UPnP" is selected under "NAT Traversal".

Click "Next" and "Finish" to return to the main menu as the other options are best left unchanged. Hit the "Close" button and that is it for the basic configuration. All the other settings are optional and don't have to be changed for the VoIP provider to work with AGEphone Mobile. Please feel free to take a look in the settings manual if you need an explanation for a specific option.

I hope this little tutorial helps you to get your internet telephony provider up and running. If after reading this you still have trouble getting things to work please always feel free to leave a message in our the forums, chat with me here, drop me a mail or take a look at our support contact page.