Frequently Asked Questions (FAQ)

General FAQ

1. What is VoIP?
2. How does VoIP work?
3. Why use IP for voice?
4. What is the relationship between codec and VoIP
5. What advantage does VoIP can provide?
6. What is the difference between H.323 and SIP
7. What do I need in order to use SIP?
8. How do I make a call?

Advanced FAQ

1. What is speech quality?
2. How are voice quality normally rated?
3. What is codec?
4. What is the relation of codec and VoIP
5. Which codec should I choose?

1. What is a difference between a free and pay version of SJphone?
2. What is the difference between SJphone versions for UNIX, MAC, Windows, and Windows CE?
3. What is an SJphone profile?
4. Can I use several soft phone profiles?
5. Can I make my own soft phone profile?
6. When SJphone establishes a call, only one party can hear the voice.
7. Voice has bad quality. Help!
8. How to get the best sound quality?
9. Can SJphone work with a USB handset, blue tooth headset, or other audio devices?
10. SJphone microphone and volume sliders do not move, and/or sound from other parties is bad, but they can hear me quite well. My sound card is based on the High Definition (HD) sound technology.
11. Is SJphone compatible with Windows XP Service pack 2?
12. SJphone cannot connect to my ITSP while my Windows CE device is on its cradle.
13. I have registered SJphone on the SJ Labs site, but have not received its registration key.
14. I forgot my registration password.

General FAQ

1. What is VoIP?

Voice over IP is an emerging technology based on open standards of IEEE, fundamentally the Internet Protocol, that allows voice data to travel across the Internet.  There are many method to used this technology, the most common and well known are SIP, and H.323.

2. How does VoIP work?

Basically VoIP is a technique to send voice information in digital form in discrete packets over digital network rather than by using traditional circuit switch (PSTN).  To do so we will need an analog to digital converter on sender side to translate the voice (analog signal) to digital than transmit it, and on the receiver end it will also need an analog to digital converter to covert the digital signal back to analog to the person being called can heard the voice.

3. Why use IP for voice?

Traditionally telephony carrier use circuit switching for carrying voice traffic.  As circuit switching is designed to carry voice and it does it very well.  Than why use IP for voice?  As broadband booms, and technology evolve.  People now want to communicate  through  various way not just voice such as email, instant messaging, video and so on.  Traditional telephony can not evolve as quickly as the demand and develop new feature on circuit switch takes much time and money.  IP is an already exist standard and many type of service already runs on IP, by using IP as a platform integrate service is now possible and low cost where traditional circuit may take long time to achieve.

4. What is the relationship between codec and VoIP

 In order to transfer voice (analog signal) over IP it first need to be digitized.  Codec is a technique to digitize analog signal to digital and vice versa.  There are various speech codec available and can be used with VoIP each with it's advantage and disadvantage.

5. What advantage does VoIP can provide?

The advantage of VoIP is it can provide advance services such as joining e-mail, instant messaging, video, voice mail all together.  Where current circuit switching (PSTN) can not.

6. What is the difference between H.323 and SIP?

H.323 and SIP both support VoIP and multimedia communications. H.323 is an older standard developed by the ITU. A good chunk of it is based on ISDN which comes from the traditional telephony world. H.323 is a binary protocol and is fairly complex in nature. SIP was developed by the Internet Engineering Task Force (IETF) and is text based (similar to HTTP). Much of the infrastructure already in place to support HTTP has been adapted to support SIP.

7. What do I need in order to use SIP?

To use SIP you need compatible software or equipment that handle the SIP protocol you will also require a high-speed (broadband) Internet connection (such as DSL or cable) for good quality calls. SIP equipment include handsets and adapters commonly referred to as hard phones. Software are numerous and include software like xlite or sjphone etc to mention but a few also commonly referred to as soft phones. You may use an adapter (ATA) to convert your normal telephone to communicate on the internet and thus make VoIP calls. The adaptor can connect to any available ethernet jack (RJ-45) on your home or business network.

8. How do I make a call?

Calls fall in four categories, local, TTCL network, mobile network and international calls.

1. By local call we mean calls to fellow users on the same SIP PBX, these calls are free.
2. Calls to the TTCL network use this format; City Code + Phone number.
3. Calls to the mobile network meaning Celtel, Tigo and Vodacom; one simply dials the number as is.
4. To make an international call simply dial the phone number using this format:
Country Code + City Code + Phone number

Advanced FAQ

1. What is voice quality?

Voice quality is how well an person can hear the voice on the opposite end.

2. How are voice quality normally rated?

Voice quality is most commonly rated through a voice quality metric called the Mean Opinion Score (MOS) which is recommendation by ITU-T.  The  MOS is a 5 point scale where 5 represent excellent voice quality and 1 represent bad voice quality.

3. What is codec?

Codec is a algorithm which converts analog signal into digital signal and vice versa.  There are three main type of waveform codec, source codec, and hybrid codec. Each consume different amount of bandwidth and provide different voice quality level.

4. What is the relation of codec and VoIP

As VoIP is a general term send voice information in digital form in discrete packets over digital network and this digital network is public network, thus there maybe other packet such data packet uses network at the same time.  The codec choose is related to how much bandwidth voice packet will consume.  In bandwidth wise aspect the smaller amount of bandwidth used the better.  But in voice aspect the higher quality the better. 

5. Which codec should I choose?

As which codec choose is depending on what codec is supported on both end of the VoIP host.  Generally a codec with low bandwidth consumption and high voice quality is a good codec .

User FAQ

1. What is a difference between a free and pay version of SJphone?

The answer is simple: there is no difference, except the G.729 codec. Both have the same free codec set: G.711, GSM, and iLBC, which are enough even for a dial-up connection.

2. What is the difference between SJphone versions for UNIX, MAC, Windows, and Windows CE?

A.: Basically they are the same, but some difference may exist due to delay in our software development process. We are working on synchronization, but some new features may appear in some releases a week or two ahead of other releases.

3. What is an SJphone profile?

A: It is a set of SJphone settings for a particular internet telephony service provider (ITSP) or other internet telephony calls. For internet calls and for calls to / from regular telephones, you need to purchase an account with ITSP. Install a new soft phone profile, initialize it with your login and password, and enjoy! Also you certainly can make direct host-to-host calls using PC-to-PC SIP or H.323 profiles for free.

4. Can I use several soft phone profiles?

A: You can sign up to any number of different services and easily switch between them.

5. Can I make my own soft phone profile?

Yes. SJphone works fine with most ip-PBX, SIP-proxy, H.323 gatekeepers and gateways, so experienced users may create new profiles themselves or even build their own IP-telephony network. If you like it, purchase a customized SJphone from us!

6. When SJphone establishes a call, only one party can hear the voice.

A. That may be because of two reasons. First, a firewall or Network Address Translation (NAT) may exist between two parties. In this case, your ITSP should support a special technology to traverse NAT, called STUN and TURN. Consult its technical support for details.
The other reason is an old half-duplex sound card driver. Please upgrade it to a full-duplex one (it should record and play simultaneously: to be sure try to run a recorder and a player at the same time).

7. Voice has bad quality. Help!

A. Some cheap or old sound cards and microphones can produce a very low quality of sound. Another reason is packet losses on the Internet due to a bad connection, a narrow or shared bandwidth. The cure for a narrowband connection is a narrowband codec: GSM, iLBC, or others.

8. How to get the best sound quality?

- Most notebooks and WLAN-PDAs have an embedded microphone and speakers. To make sure that your microphone is good enough try to record your speech;
- To get better audio quality and to avoid echo, you may use any headset, USB phone (download a driver from our web site), Blue Tooth, any wired headset etc;
- We recommend you to install the latest updates of operating system, sound and network drivers, and DirectX (check with dxdiag.exe, it should be 7.0 or higher). Get updates from the vendor web site or just run the MS Windows update;
- You may have problem with old consumer WLAN-PDAs which were not designed for internet telephony. Most of the latest devices have embedded wireless, good CPU and sound system now, and work perfectly;
- Check your Internet connectivity: try to ping your party's IP address. A good response time is below 100 ms unless the party you are calling is from Down Under. If the ping time seems too large, you Internet connection is too busy now. Try to stop any other activity. If the problem persists, ask your Internet provider or system administrator to fix it.
- For the best voice quality, use the G.711 codec on broadband, on dialup try the GSM, iLBC, or other codec's;
- Normally you should hear Voice over IP speech much better than using an old plain or mobile telephone. A delay, an echo, a sound drops, and a distortions show network or hardware problems.

9. Can SJphone work with a USB handset, blue tooth headset, or other audio devices?

A. Generally, SJphone works with any devices which Windows detects as audio hardware.
To use it, go to Options -> Audio -> Sound Devices and select it in the Playback and Recording.
Support for other functionality like dial pads, dial/hang up buttons, and ringers varies, because each device has its own interface, quite often incompatible with others.
The list of supported USB handsets is on the Support page (Download Windows XP/2000 drivers for USB phones, http://www.sjlabs.com/sjp.html). Sometimes USB handsets are being sold under various brand names, and we cannot trace all that names.

10. SJphone microphone and volume sliders do not move, and/or sound from other parties is bad, but they can hear me quite well. My sound card is based on the High Definition (HD) sound technology.

A. Go to the Audio tab and disable the Enable Direct Sound for improved audio performance option.

11. Is SJphone compatible with Windows XP Service pack 2?

A. We have tested SJphone on machines with XP SP 2 installed and found that it works normally with minimum required adjustments in the Windows Security Center. We have tested direct SIP and H.323 PC-to-PC calls, and calls through H.323 Gatekeepers and Gateways, and SIP Proxies.
Below are our recommendations:
If you turn off the Windows firewall in the Security Center, no settings are required to change for Windows or SJphone.
Windows Security Alert Message
If XP Service Pack 2 is installed on your computer with its default-enabled internal firewall, the Windows Security Alert message will appear the first time you start SJphone. This message will inform you that it has blocked SJphone from accepting connections from the Internet or a network. To allow SJphone receive incoming calls, click the Unblock button on the message. Now SJphone will be able to receive incoming calls. No other additional settings are required for Windows or SJphone.
If you have clicked the Keep Blocking button, SJphone will not be able to receive incoming calls. To allow SJphone receive incoming calls,
* Check that SJphone is running.
* Go to Security Center and click Windows Firewall.
* Check that Don't allow exceptions is not selected on the General tab.
* Go to the Exceptions tab and select the SJphone.exe checkbox.
* Click the Edit and then Change scope buttons and check that the Any computers (including those on the Internet) option is selected.
* Click the OK button.

12. SJphone cannot connect to my ITSP while my Windows CE device is on its cradle.

A. When any WinCE device is in its cradle and connected to a host PC via Active Sync, SJphone cannot connect to any host outside its local network. In this state, SJphone cannot register with any service, proxy, gatekeeper, make any calls to other soft phones., etc. , which is outside the local network, even if SJphone shows "Ready to call" in the Simple mode of its interface. If you switch to "Advanced mode" (Menu->To Advanced Mode), SJphone will show NAT/Firewall: Blocked. This issue is connected with Active sync To make calls outside the local network, you should take your WinCE device out of its cradle and possibly restart SJphone. (Menu->Restart)

13. I have registered SJphone on the SJ Labs site, but have not received its registration key.

A. Assuming you have entered a valid e-mail address for the registration, most likely some spam detection software decided that the e-mail with the registration key is a spam and killed it.
To obtain your registration key directly from SJ Lab web site,
* Go to Options -> Support tab and click the Register button
* Click the Get Key on the SJphone registration message box
The Product Registration page will appear in your default web browser.
* Enter your e-mail address used for the registration as Login, then your password, and click the Logon button. .
The Main Menu page will appear in your default web browser.
* Click All registered products and see the list of all your registered products with their registration keys.

14. I forgot my registration password.

A. If you forgot your password,
* Go to Options -> Support tab and click the Register button
* Click the Get Key on the SJphone registration message box
The Product Registration page will appear in your default web browser.
* Click I forgot my password.
The Password Retrieval page will appear in your default web browser.
* Enter the e-mail address to which you have registered SJphone.
An e-mail with the password will be sent to the e-mail address used for registration and the Your password has been sent to your e-mail address page will appear in your default web browser.