Frequently Asked Questions (FAQ)
Jump to a question:
Why would someone use EzVoIP.co?
Different people, different rhythms. Here's the top 10 reasons to use our services:
You want to save money while not compromising voice quality and features.
Concerned that cell phones emit harmful radiation, you want to cut down mobile phone use while keeping or increasing your connectivity and paying as much as now if not less.
You feel that the cell phone company is ripping you off (Canada, for instance, has the most expensive mobile phone service in OECD - CBC, Geist, Rabble, Sun, OpenMedia, G&M, Tee) but don't want to give up using your cell phone - just pay less while getting more options.
You travel often and want to take your phone number and existing plan with you without paying outrageous roaming charges.
You find that with regular phone companies a landline is even more expensive than a cell phone.
You use a card to call other countries but find that you never speak for the minutes they promise you, paying much more than the advertised rate. With us, you can almost immediately see in your Control Panel how long you talked and how much you were billed, almost always in 6-second increments.
You need advanced capabilities (IVR or calling menu, multiple / unified voice mailboxes, multiple phone numbers at less than $2 / month, calling card, toll free number, foreign phone numbers) and require features other providers cannot offer you for free.
You have a relative who wants to keep using their landline, they now pay more than $10 / month and you know that their phone company is taking advantage of them.
You want an easy way to control telemarketers, anonymous callers or calls during your sleeping hours.
You want to simplify your life and stop handing over your money to monopolists or large soulless conglomerates.
TCP or UDP?
When setting up your softphone or device you typically have to choose between TCP and UDP. These are two transportation protocols, the chief difference being that UDP is more forgiving with lost packets and will just "move on", whereas TCP might not. This might suggest that UDP is a better choice, given the nature of SIP (VoIP) communication, but the reality is a bit more complex, because TCP is more "lightweight" and needs less "keepalive", saving battery on mobile devices. See StackOverflow for more details.
Who is Lenny, telemarketers' and annoying callers' endgame?
Lenny is a set of prerecorded voice messages that kick in sequentially whenever there is silence on the other end, creating the appearance of a conversation, designed to waste the caller's time. We can send certain callers to Lenny based on Caller ID or failure to perform a simple test, such as pressing a digit on the keypad within the allotted time as part of an IVR.
If you have an account with us, you may also transfer an annoying caller straight to Lenny. You could say, for instance, Please hold, I'll transfer you to Lenny, our "vice-president" (or, if it's a residential number, "the head of our household").
Here's Lenny talking to a Vonage telemarketer:
YouTube Video - Recording of Vonage Telemarketer
Here's Lenny talking to a Wildrose Party telemarketer:
YouTube Video: Wildrose party telemarketer
Here's Lenny talking to a Progressive-Conservative (PC) Party person campaign volunteer soliciting donations (she's actually very patient!):
YouTube Video: PC Party / Polievre
The last recording, above, was covered in the Canadian press:
<<Nice fellow, that Lenny, though as a campaign worker for Pierre Poilievre discovered recently, you don’t want to get stuck talking to him.
“Oh good. Yes … yes, yes,” answers Lenny when the worker, whose name sounds something like “Segalle,” asks if he’s willing to put an election sign on his lawn.
But as a recording of the 11-minute, 14-second conversation posted to YouTube reveals, it’s all befuddlement from there.
“What was that again?” Lenny asks in his soft British accent. “Sorry, again?”
Segalle obligingly raises her voice as she quizzes Lenny on the voting intentions of his household, only to be treated to inside information on Lenny’s daughters — Rachel, a bit of a hothead, you know, and Larissa, the third-eldest but first in the family to go to university — before he again asks her why she’s calling.
And then there are the ducks. “Sorry, could you just hang on for one second here, hang on,” says Lenny to a chorus of quacks in the background.
Lenny, as you’ll have gathered by now, though poor, patient Segalle apparently never did, is a recording — a series of pitch-perfect responses designed to fool callers into thinking they’ve reached a genuine person with genuine interest in whatever they’re pitching.>>
Technical Details
Here's Lenny's DNA (FreeSwitch LUA Code, also at http://pastebin.com/ujUzxsxz):
--[[
It's Lenny in Lua
--]]
local path = '/usr/local/freeswitch/sounds/lenny/';
local counter_mainloop = 1;
local counter_noinput = 1;
local counter_consec_noinput = 0;
function get_filename (aType, aNumber)
local file = path .. aType .. '/' .. aNumber .. '.raw';
local fd = io.open(file, "r");
if fd == nil then
file = path .. aType .. '/' .. aNumber .. '.gsm';
end
io.close(fd);
return file;
end
function wait_for_silence(aTimeoutIncrement)
if aTimeoutIncrement == nil then
aTimeoutIncrement = 2000;
end
freeswitch.consoleLog( 'DEBUG', "ENTER wait_for_silence " .. aTimeoutIncrement .. "\n" );
if session:ready() ~= true then
return false;
end
session:setVariable("wait_for_silence_timeout" , "");
session:setVariable("wait_for_silence_listenhits", "0");
session:setVariable("wait_for_silence_silence_hits", "0" );
session:execute( "wait_for_silence", "300 30 5 " .. aTimeoutIncrement);
local timeout = tonumber(session:getVariable("wait_for_silence_timeout"));
local speech = tonumber(session:getVariable("wait_for_silence_listenhits"));
local silence = tonumber(session:getVariable("wait_for_silence_silence_hits"));
freeswitch.consoleLog( 'DEBUG', "Speech : " .. speech .. " Silence : " .. silence .. "\n" );
if speech > 20 then
wait_for_silence( aTimeoutIncrement );
return true;
else
return false;
end
end
function play_next_mainloop ()
session:execute( "playback", get_filename( 'mainloop', counter_mainloop ) );
counter_mainloop = counter_mainloop + 1;
local fd = io.open( get_filename( 'mainloop', counter_mainloop ) , "r");
if fd == nil then
counter_mainloop = 1;
end
counter_consec_noinput = 0;
io.close(fd);
return 2000;
end
function play_next_noinput ()
session:execute("playback", get_filename( 'mainloop', counter_mainloop ) );
counter_noinput = counter_noinput + 1;
counter_consec_noinput = counter_consec_noinput + 1;
local fd = io.open( get_filename( 'noinputloop', counter_noinput ) , "r");
if fd ~= nil then
counter_noinput = 1;
end
if counter_consec_noinput > 3 then
counter_mainloop = 1;
end
io.close(fd);
return 3200;
end
session:answer();
session:execute( "playback", get_filename( 'greeting', 1 ) );
local SilenceTimeout = 4000;
while session:ready() == true do
if wait_for_silence( SilenceTimeout ) ~= true then
SilenceTimeout = play_next_noinput();
else
SilenceTimeout = play_next_mainloop();
end
end
A loop code was published:
[Lenny]
exten => talk,1,Set(i=${IF($["0${i}"="016"]?7:$[0${i}+1])})
same => n,ExecIf($[${i}=1]?MixMonitor(${UNIQUEID}.wav))
same => n,Playback(Lenny/Lenny${i})
same => n,BackgroundDetect(Lenny/backgroundnoise,1500)
..together with a ZIP archive.
Make Lenny part of your IVR, as heard and discussed; also, install.
What is a server, domain or POP and how do I choose it?
A server may interchangeably be called domain or POP (Point of Presence); they are sometimes defined as "dial-in nodes of Internet Service Providers through which the customer gains access to the Internet".
When signing up for our service we recommend a server based on what we approximate to be the closest server to you. If you have a phone number (for incoming calls), that number (aka DID) is assigned to a server, and unless you are connected to that particular server, you will not be able to receive incoming calls - they will all go straight to your voicemail, without ringing. However, if you are travelling, connecting to your home server might be impractical due to latency. In that case, you could order a new phone number, which may very well be temporary, in your new area. We could then attach it to the server closest to you and forward the calls from your home phone number to the new temporary number. You can test yourself, objectively, which server is best for your new location.
Back when we supported "ping", you could run such a test on each of our servers in your vicinity. Our servers follow the format [city]#.ezvoip.co so that for example the Toronto server would be toronto1.ezvoip.co. We do not have servers in every city in the world, but we do have them in most major North American cities as well as a few European and even Australian ones and we're constantly expanding our presence.
For security reasons, we currently block ping on our servers. Henceforth, please consider looking at the list of servers and pick the one geographically closest to you. If you don't know the distance, you may find out using your favorite search engine.
You may run such a test on virtually any platform and operating system.
Where are your servers (aka domains, POPs) located?
We have VoIP servers located mostly in USA and Canada (New York, Los Angeles, Houston, Dallas, Montreal, Toronto), but we are rapidly spreading in the rest of the world: London-UK, Paris-France, Melbourne-Australia. We will generally recommend or choose the server closest to you automatically, based on the address you provide when subscribing. If using an Outgoing Only plan, when traveling, it is best to use the server in your city or in the city closest to you. While it is possible to use a far-away server with good results, for best performance you might want to determine which is best for your location (which may not necessarily be the geographically closest server, though it usually is).
If you have a phone number, it is associated with a particular server ("home server") and if you connect to another server, you will not be able to receive calls as they will go straight to voicemail. If traveling extensively, it is best to order a new phone number in your new, temporary location and forward the calls from your home phone number to your new number.
The server which provides you least latency should be used. Also, please note that as we expand our presence, we update the list and not so much the map image.
In summary, our international servers are in Amsterdam, London, Paris and Sydney (typically only one server in each city). Our servers take the form city#.ezvoip.co so if there is only one server, it would be paris1.ezvoip.co. In North America, we have Atlanta (2), Chicago (4), Dallas (2), Denver (2), Houston (2), Los Angeles (4), New York (8), San Jose (2), Seattle (3), Tampa (4), Washington (2), Montreal (10), Toronto (10), Vancouver (3). We also maintain an explicit list.
Why doesn't my smartphone/softphone ring on incoming calls?
Some of our customers had difficulties with incoming calls on their smartphone softphones or even Windows, and more specifically, with the great Zoiper.
This article deals with the problem of being able to make outgoing calls but not receiving incoming calls. You can test outgoing calls by calling our echo service at 4443. If that works, it means that you have a working connection to our server and are able to place outgoing calls.
If you can place outgoing calls but cannot receive calls, one simple reason may be that your DID (phone number) is set up on one server (e.g., Toronto5) and you connect to a different server (e.g., Toronto2). For incoming calls to "ring" you need to connect to the same server as your DID. You should always connect to the server closest to you geographically, and if there are several in the same city, pick one of the higher numbers, then adjust the DID from your portal to use that server or POP (point of presence).
Another very simple reason why you might not be able to receive incoming calls is that you have closed or killed your softphone. A softphone will not ring if it is not working.
If you are certain that you are connecting to the same server as your DID, but still cannot receive incoming calls, there is most likely an issue with your LAN or your device (smartphone or laptop) blocking connections to your router and thus to our server. In particular, your OS might "kill" your softphone program or its connection after a while for power saving or security reasons. For example, some Windows computers will disconnect Internet connections when the screensaver kicks in. That would definitely prevent incoming calls from ringing.
Even with updated instructions, softphone developers change their interface and create new features and GUIs all the time, which is why specific instructions are not always useful - it is better to have a general understanding of what we are trying to accomplish. The following tips might help in setting up softphones for incoming calls.
Most softphones on smartphones (such as Android, iOS, Windows Phone and Blackberry) are designed to conserve battery and for low resource use. Maintaining a connection for incoming calls uses more battery than having a connection "on-demand" only for outgoing calls. As such, by default, softphones set up accounts without incoming calls. For most softphones, receiving incoming calls requires an additional step in setup, such as checking off an "always available" checkbox. This may also be called "allow incoming calls" or "maintain connection to server".
Any number of devices may be set up with one subaccount for outgoing calls. This means that with the same credentials, you could be making a phone call to Honolulu on your Windows PC, your spouse could be making a phone call to Buenos Aires from her iPad and a relative could be calling Shanghai, all at the same time. However, if one was to call you, there is no telling which device will ring. If incoming calls are important, it is best to set up a different account for each device. You can then have an IVR menu allowing callers to choose who to call with an extension, or even have all devices ring at the same time with the Call Hunt or Ring Groups feature.
If setting up a different account for each device is undesirable (even though we can usually do so at no extra charge), you may register only one device on the server with the incoming phone number, and all the other devices, used only for outgoing calls, on geographically close but different server(s). This way, only the device registered on the server with the incoming phone number will ring (if properly setup as explained below).
In our current setup, phone numbers and their respective connected accounts must be on the same server. We generally attach a phone number (DID) to the server we recommend to you in our welcoming email and you can change the server or POP yourself from your DID options in your portal. If you decide to change the server (for instance, if you are traveling and wish to connect to a server closer to your geographical location), make sure you move your DID to the server of choice. Don't forget that you may also order a new phone number anywhere in the world, so that you can be called with only a local phone call and then forward your calls to the number connected to your other geographically close server.
Some network providers may disable incoming calls for competitive / monopolistic reasons (for instance, they might want you to use their more expensive voice service if they are a cell phone provider). In our testing, we have not found this to be the case with any of the providers we tried, yet that does no eliminate the possibility that they may do so in the future. This would violate net neutrality. See Net Neutrality by Country.
If you are using Wi-Fi, your router may not allow these connections to take place. All the routers we tested allow incoming calls with no issue, but incoming calls may explicitly be denied with a setting on some routers.
A device might disconnect from the Internet or even kill connections due to inactivity. The following are people complaining about this very behaviour with Windows 10 from 10forums, MsAnswers, Seattle. To fully monitor and troubleshoot this issue you can use ping or other tools.
Finally, when troubleshooting a partially-working connection, it is useful to bear in mind that some of the changes you make in the setup of your softphone may not be immediately reflected in the way it operates, because the softphone might not immediately update its connection, keeping the old, stale connection. Sometimes softphones update and reconnect to the server only every hour, and rebooting the smartphone or restarting the softphones may not necessarily force the new settings on. In such situations, you might need to change the "polling time" setting on your softphone and reboot, or simply wait until the next server polling has taken place.
Most importantly, be sure that the server we indicated in our Welcome Email (or updated in subsequent communication) is the server you are registering on! If unsure, delete your account in your smartphone and try setting it up again.
If you tried all the above and you are still having problems, we could try to help you troubleshoot your LAN even though that is not part of our offer, so you can contact us and provide as much information and as detailed as possible.
What is "Demarcation Point" and why is it important?
If you are using your ATA as a replacement for your regular phone line, you may want to disconnect your home wiring from the Bell lines. This is much easier to accomplish if you have a "demarcation point jack" installed, such as this (disregard the thick white cable, it is unrelated). Disconnecting your line is as easy as unplugging the jack. This ensures that you will not broadcast your conversation on the Bell (i.e., incumbent) trunk; also, you will not be receiving any interference from possibly malfunctioning Bell equipment.
Can you stop logging my calls?
We occasionally get requests from our customers to not "log" their calls on our servers. The short answer is we do not record the voice content of your calls (see our call recording answer) but we do keep track of incoming and outgoing calls (i.e., their duration and the destination of outgoing calls) for billing purposes, as disclosed in our privacy statement. We take our customers' privacy seriously and we make it easy to deal with telemarketers and unwanted callers.
The aforementioned data (call duration and destination) is called CDR, which is short for Call Detail Records. This data is generated by exchanges for billing purposes. You can find more general info about it on Wikipedia or a more technical article on voip-info. We make your CDR available to you in the Control Panel.
CDR data is essential to VoIP billing. Virtually every VoIP provider uses it for revenue generation. In EZvoip case, we lease lines and time from upstream providers and resell it to our customers. We are charged based on CDR data and we charge our customers mostly at cost (sometimes at a small loss, sometimes at a small profit, but we balance it out in the end). Without this data, we would be open to either being taken advantage of, with a client who overuses our service while underpaying, or we could be charging more than we pay our upstream provider, compromising our commitment to charge at cost while in beta stage. We also use this data to create plans that better respond to our customers' needs and to keep plan costs in sync with usage.
Encrypting calls, a feature provided with some softphones, prevents regular snooping on your conversations by third parties, much like using GPG with email safeguards your privacy. However, your provider still has the direct your phone calls to the right number, much like your email has to reach its destination, and as such, records (i.e., metadata) / CDR is generated and stored. This is true of any telephony or email provider. If an email provider were to delete such data after sending the email, or if you were to delete the sent email from the Sent folder, this data would continue to exist on servers upstream, servers that had taken the email based on its address info and sent it along. The same is true with VoIP.
We do not know of any VoIP provider to claim that they do not generate CDR data to protect customer privacy and we suspect that if any one provider makes such a claim, they are being disingenuous. While a VoIP provider could choose to not disclose CDR data to their customers, they cannot eliminate it from their upstream provider servers - and there is always an upstream provider or partner, as phone numbers are not generated or assigned out of thin air.
We will minimally cooperate with law enforcement requests - i.e., we will provide the minimum amount of data we are legally required to provide, so that we do not incur adverse consequences, but it's unlikely that law enforcement would even request anything from us since they can go to some upstream provider with whom they are likely to have a pre-existing relationship. The same is true of virtually any VoIP reseller. Not complying with such requests (e.g., a court-ordered subpoena) would jeopardize the privacy of our other customers and may even endanger our survival as a business.
To summarize our position on the collection of data, we do collect metadata (call length and destination) for billing purposes, it is stored securely and we have never sold it in the past nor will we in the future sell it to a spammer or anything like that.
If you want to make and receive calls anonymously with VoIP, with any provider, you would have to pay for it anonymously (some free services exist, but they might be logging even more data and selling it along to marketers or advertisers). Some claim that it may be possible to pay anonymously with Bitcoin. Additionally, you would have to ensure that the email address you use to register cannot be traced back to you. That is far more difficult than you would think and even familiarity with TOR does not guarantee it. Finally, keep in mind that someone with full access to your CDR and email flow metadata (that is definitely not us) might be able to ascertain (guess) your identity from that alone.
Do you accept Bitcoin or other digital currencies?
We have yet to conduct any transaction in digital currencies; however, we are open to this idea on a case by case basis. Due to higher risk, volatility and more time we have to spend in processing such a transaction, our conversion rate for any such transaction would be slightly higher than the prevailing market rate, with a 0.3 premium on the oanda.com posted rate.
Bitcoin can be used for anonymity purposes, as explained at it, simplified, mag, cd, dl. Note however, that in order to protect us and our business, we will cooperate with law enforcement to the minimum extent required by law. That includes not providing help in the commission of crime and even disclosing it to authorities if we ever become aware of it. Please use our services solely for lawful purposes and especially refrain from using us for telemarketing, spamming or automated dialers as our jurisdiction is particularly strict with such activities.
What does "Beta" mean?
We are in "Beta" and that essentially means that our presentation layer, i.e., how we present our services to customers, is not yet finalized. Although our service is rock-solid and functions even better than many other VoIP services whose ads you can see on TV, our goal is to make using the rather technical and complex VoIP paradigm as simple as using grandma's phone.
While in Beta, we do not seriously advertise and we are not making a profit - we provide services at cost. We may occasionally advertise in order to compare the impact of various strategies and the return on investment in such services, yet our client base is formed almost exclusively of people who have specifically searched for excellent service without compromising quality.
We pay taxes on the services we resell and our pricing includes this cost, which others may charge separately or may not collect at all. Additionally, we incur webhosting and domain hosting costs. We do not get our telephony services for free, we get them from an upstream provider. Our markup reflects taxes, PayPal and conversion rate charges as well as domain and webhosting costs. While in Beta, we do not charge for our time. We often end up selling at a loss.
We will consider ourselves to be out of the Beta stage once we are satisfied that our service is sufficiently easy to use, at a cost that compares favorably to our competition and / or we have a special and in-demand feature or service that very few competitors or nobody else can provide.
Call Recording with EZvoip
People record phone calls for a myriad of reasons. Some may want to record phone calls to help remember important commitments and promises, to keep their word or to help others keep theirs. Other times, one might want to ensure that they have proof of illegal deeds or threats. Either way, you should know that "secretly" recording a telephone conversation is legal in many parts of the world and illegal in others. We cannot provide legal advice as to what is legal and what is not, as this Wikipedia article on telephone recording laws does. We are providing here ways to record telephone conversations with the caveat that it is your responsibility to check the legality of recording such conversations in your jurisdiction and obtain full and informed consent wherever that is required.
At EZvoip.co, we offer primarily email-based support and as such we do not have a need for call recording, but we might sometimes record calls made to us for the purpose of training or improving service. Irrespective of what we do, law enforcement, various other government agencies and even upstream providers may record telephone conversations without our or our clients' knowledge and / or permission.
Following customer requests, we started beta testing the ability to automatically record phone conversations server-side in 2020. We expect to be able to provide this service at an additional $0.003/minute (i.e., less than a third of a cent per minute). Such recordings would be stored on our server and we are working to make them available through our regular interface. We may have to charge extra for manually searching and providing recordings. We recommend that if you need this feature, you do client-side recordings and use our (server-side) service as a backup, at least until we have a more permanent solution. We do not have an ETA for finalizing this feature.
We only enable this service (server-side recording) on request.
By requesting and using this service, you acknowledge and agree to be solely and entirely responsible for complying with all international, federal, state, provincial and local laws and regulations in the relevant jurisdiction when using this feature, and you warrant to EZvoip.co that you are acting legally. You expressly acknowledge and agree that EZvoip.co, its providers and associates will not and cannot be held liable for any and all claim, directly and indirectly related to your recording of phone conversations. Without limitation to the Terms of Service, you agree to release, indemnify and to hold harmless EZvoip.co, its providers and associates from and against any and all claims, damages or liabilities of any kind related directly or indirectly to the recording of any phone conversation using the Service.
To warn your callers, you might want to have a recorded message. You can create your own recording or you can send us your text your choice of male/female voice and we can generate the proper recording for you at $5/file (max 30 seconds). Such a warning is typically played at the beginning of a Queue, IVR or before a Ring Group. You can do such a setup yourself or we can do the first one free, with subsequent modifications at $3 each.
If you are using your cell phone company services, on Android you can try using Call Recorder (4 - 192073), Automatic Call Recorder (4.2 - 623284), Call Recorder (4.3 - 144174), Super Call Recorder (4.3 - 11729), Boldbeast Call Recorder (4.0 - 2870), Contacts Phone Dialer (EC 4.6 - 125907), while on iPhone you have Call Recorder (4+ - 58), Tape-A-Call Pro ($10 4+ 1454), NoNotes (20min free/month, 4+ 300), IntCall ($0.10/min 4+ 824). You may also record iPhone calls on your Mac or MacBook via FaceTime with Audacity. For Skype, which is a proprietary VoIP service, you have MP3 Call Recorder and VodBurner. On Google Voice, with a proper setup, you may press 4 and a prerecorded notification will be played to both you and the other party, and it works only for incoming calls; you may circumvent the latter restriction via GetHuman. We indicated the rating and number of users leaving feedback at the time of writing - these numbers will definitely change in time. (The above list is not kept up to date.)
The problem with the aforementioned recording methods is that they are either very expensive, or they tend to NOT work in different ways, as a perusal of user feedback on these applications reveals. Some require a "rooted" Android or "jailbroken" iPhone, others route your call through different countries and save your recording remotely, compromising your privacy, others do not record the remote party, only yourself, and finally, some only work with some devices and not with others. Those that do work may have significant limitations and may require you to pay significant amounts of money in order to access your recordings.
A not-so-obvious trick employed by some of the aforementioned solutions is that they will route your call via a VoIP connection, which is what we are providing by default. This is an implicit recognition of the fact that voice call recording is best done via a VoIP connection as it is far easier to record a stream that is already made available at application level, without all the OS "security" hoopla.
With SIP VoIP, you may record a phone call "client-side" in many free softphones and your call and recording is not routed anywhere else, it is saved on your smartphone or computer, locally. This is provided in Linphone on most platforms for free, on Zoiper in their Android and iPhone versions and as a paid feature on Windows and iOS. You may also record for free with Xlite on Windows and CSipSimple or SipDroid on Android. MicroSIP is probably the simplest OSS Windows-only solution; Blink is a good choice for MacOS primarily, but also Windows and Linux as it is also OSS. Some non-free apps start by offering this feature gratis, but in subsequent versions it is migrated to the paid version; for this reason, we recommend OSS (open source software) because you usually get the feature free of charge with no limitations.
It may also be possible to record the phone conversation for free with the afore-linked Audacity, on desktop platforms such as Windows, Mac and Linux. For Windows, we recommend the Portable version.
The open nature of our service allows a wide variety of recording options that give you complete control and privacy. This is much more difficult with plain-vanilla cell phone services.
How to best use EZvoip.co with a Smartphone?
Many, if not most of our customers are using our service with a smartphone. Though we welcome this development, use with a smartphone presents a unique set of challenges. In particular, making IP-calls across 3G and WLAN does not work well if you are moving around or have a weak signal, and this is not a limitation of our service, but rather a limitation of the technology; such calls don't usually work well with your cell phone provider either.
First, you probably want to install and use a SIP client for VoIP. Linphone Zoiper is our recommended client for virtually all mobile platforms, including Android, iTunes, Blackberry and Windows Phone. You may find additional clients as follows:
If you have an Android phone, search for Android App Market (Google Play): SIP VoIP Free. Good, free choices are CSipSimple or SipDroid; 3CX can interact with more complex setups while the great Zoiper adds Instant Messaging capabilities.
If you have an iPhone, searching iTunes App Market finds some common apps, such as, again, Linphone and Zoiper. Other apps are Join and 3CX. Siphon, though completely free and ad free, may require jailbreaking.
In particular, the Internet connection will always be wireless and as such inferior to a wired connection. Though it is possible to have a flawless conversation over WiFi, for certain setups, that is more the exception rather than the rule. If you are using a "Data / 3G / 4G" connection - i.e., the Internet you get from your cell phone provider - your connection might be intentionally degraded or limited by your provider, but it is often times possible to use it. Incoming calls may be harder to set up.
One simple way to increase the quality of your connection is to use wired headphones and keep the phone away from you, in a position where it gets the strongest WiFi signal. This will also save your brain from being "microwaved" even though VoIP over WiFi or even the Data / 3G / 4G connection is not as bad as most mobile voice connections. As always, it also helps if the Internet connection is not saturated (overused) and the router is configured for QoS. An old router may also cause the connection to suffer. Finally, quit and remove from the memory of your phone all programs you are not using, so that the SIP / VOIP program you use to make phone calls has all the resources it needs.
You may also be able to achieve a marked improvement in connection quality by finding the best codec for your situation. (A "codec" is the compression scheme used to "pack" voice data for transmission over the Internet.) We are offering currently G.711U, G.729A and gsm - various softphones may use slightly different names for them: for instance, G.711U may sometimes be called "u-law", "mu-law" or "μ-law". G.729A is usually expensive and not implemented in free applications, and while it manages to compress voice data very well, the quality may be unacceptable to some. GSM compresses data even more to the point where there is a noticeable decrease in quality. G.711U offers perhaps the best and most natural voice quality and we recommend it, if you have the bandwidth. If you are not happy with voice quality, we recommend you force the use of G.711U or u-law and disable GSM.
If you find yourself often using our services somewhere where a wired connection is possible - for instance, at home or at the office, consider setting it up on a desktop computer, laptop or tablet. Better yet, using a dedicated fixed device such as an ATA, connected to your router, with a corded or cordless phone is perhaps the most relaxed and comfortable use of VoIP - it's as reliable as a landline phone, but without the expense and with better voice quality. Ask us about it, we might be able to provide it for free.
Enjoy!
What is QoS (Quality of Service)?
Any properly configured VoIP system can and should provide a voice quality that is better than regular POTS / PSTN (old copper wire phones), not to mention cell phones. Consumer grade VoIP service is, unfortunately, seldom properly configured.
Better voice quality can be achieved in two ways. First, make sure that your the uplink of your asymmetrical connection is not eaten up, as explained at Consumed Consumer. You will know you are having an uplink problem if you can hear the other party well in VoIP conversations but they complain you are breaking up.
Secondly, make sure that you configure your router to give priority to VoIP calls via Quality of Service (QoS) settings, as shown below:
If your router does not allow you to implement such settings, you might still be able to install an open source firmware that unlocks such settings, such as DD-WRT.
What is a codec and which do you support?
The codec in use with VoIP communications refers to the algorithm used to compress and encapsulate data so that it can travel over the Internet and be reassembled with minimal disruption and best possible voice quality. Different codecs exist to accommodate a wide variation of conditions, such as bandwidth and processing power of the equipment doing the real-time compression/decompression.
The codecs we support are G.711 (μ-law / pcmu) , G.729 and GSM. Here is their bit rate and bandwidth requirements, both in Kbps:
Codec
Bit Rate
Bandwidth
G.711u
64
87.2
G.729a
8
32.2
GSM
13
29.2
See also our recommendation.
How much does VoIP cost over mobile Internet (aka Data plan)?
While costs vary and technologies change, codecs will likely use the same amount of data. The following was quantified with 3G service; while service levels improved, faster service may cost more or less the same.
If you have a smartphone or iPod, you can use VoIP also over wireless Internet at home, and since that is cheap and unmetered, the cost would be zero. When using it on the go, you have to use the 3G Internet connection from your cell phone provider which is metered. The cost will vary with the bit rate, itself determined by the codec in use:
Codec MB/min cents/m
G.711u 1 5-10
G.729a 0.5 2-5
GSM 0.5 2-5
If you use exclusively G.729a (recommended), your cost could be as low as $0.02/min (depending on your plan). You would have no roaming charges to worry about and calling overseas would not mean you have to take a second mortgage. This compares favourably to a $0.10-25/min which is what you end up paying for voice on mobile phones in various plans.
You could easily get a $10/month voice plan + $20/month for data (which usually includes 500 MB - 1 GB). At a rate of 0.5 MB / min that translates in 1-2000 minutes of VoIP. You can also use your phone's 3G to send out and receive free SMS, bypassing your cell phone company altogether.
Finally, if financial reasons are not sufficient for you, consider the repeated warnings from various bodies including the World Health Organization and, more recently, Health Canada suggesting that cell phone use is "possibly carcinogenic" and must be reduced, the conclusion is simple: minimize your use of voice over cell phone and, if you still need to talk to others, maximize your VoIP usage the EZ way!
What is the best codec to use?
All codecs offer some kind of compromise between sound quality (which is a rather subjective attribute) and resources used (bandwidth and processing power). The best codec for you may not necessarily be the best for your conversation partner.
Most people find that the most natural sounding voice is provided by G.711u, also known as u-law, μ-law or pcmu. However, you can still maintain an pretty good voice quality and lower bandwidth usage with codecs like G.729. Note that GSM has the worst voice quality but offers the best bandwidth savings. If you specifically require it, please let us know.
Codecs such as GSM and G.711u are usually provided for free in free softphones, but G.729 may be available only at extra cost due to licensing issues. Since it uses more bandwidth than the free GSM while providing only a negligible improvement in voice quality, we recommend you do not use G.729. If bandwidth is not a concern (and with most modern computers or Data plans it is not), we recommend you disable all codecs except G.711u (or ulaw).
See also what is a codec and which we support.
Why would someone use EzVoIP.co?
Different people, different rhythms. Here's the top 10 reasons to use our services:
You want to save money while not compromising voice quality and features.
Concerned that cell phones emit harmful radiation, you want to cut down mobile phone use while keeping or increasing your connectivity and paying as much as now if not less.
You feel that the cell phone company is ripping you off (Canada, for instance, has the most expensive mobile phone service in OECD - CBC, Geist, Rabble, Sun, OpenMedia, G&M, Tee) but don't want to give up using your cell phone - just pay less while getting more options.
You travel often and want to take your phone number and existing plan with you without paying outrageous roaming charges.
You find that with regular phone companies a landline is even more expensive than a cell phone.
You use a card to call other countries but find that you never speak for the minutes they promise you, paying much more than the advertised rate. With us, you can almost immediately see in your Control Panel how long you talked and how much you were billed, almost always in 6-second increments.
You need advanced capabilities (IVR or calling menu, multiple / unified voice mailboxes, multiple phone numbers at less than $2 / month, calling card, toll free number, foreign phone numbers) and require features other providers cannot offer you for free.
You have a relative who wants to keep using their landline, they now pay more than $10 / month and you know that their phone company is taking advantage of them.
You want an easy way to control telemarketers, anonymous callers or calls during your sleeping hours.
You want to simplify your life and stop handing over your money to monopolists or large soulless conglomerates.
What special numbers do you support?
4443: Voice Echo Test (will speak back to you – to test how others hear you)
4747: DTMF Echo Test (to ensure that dial tones are recognized)
1-555-555-0911: Test CallerID and e911
044+Country Code+number: International Premium (override account setting for better quality at higher rates)
*97 Voicemail (*98 for prompt - if you have more than one)
311: Non-Emergency Police or Municipal Services (Canadian Servers)
511: Weather and Traffic updates (Canadian Servers)
811: Non-Urgent TeleHealth (Canadian Servers)
822: Caller ID read-back
We typically increase the range of such numbers we support but we may lag in updating this list; please be sure to email us to let us know if you have a specific request.
What are some special recordings?
We were often asked what are some of our special recordings and how can they be reached. Here is a list of the dialing codes for such recordings. You may use them by the dialing code in IVR call flows as well as other features such as in Caller ID Filtering.
06812984 - 5s silence
0682811 - 25c
0682818 - asleep
0682812 - coin
06813224 - N/A
0682810 - call traced
06838157 - O-Zone
0682808 - outgoing only
06839979 - please visit ca
0682806 - sorry - cow laughing
06859060 - sorry lenny throne
0682807 - number disconnected
06813228 - win balloon
06813228 - win info
Be sure to let us know if you have further suggestions!