ZyXEL V100 (V100 Softphone 1 Runtime License) Support Notes Version 1.00 April 2009 1
Contents Overview 1. Overview of V100 Softphone...3 2. Setting up the V100 Softphone.....4 3. V100 Basic Phone Usage.....7 3.1 Making a Call........ 7 3.2 Receiving a Call...8 3.3 Ending a Call...8 3.4 Placing a Call on Hold..9 3.5 Transferring a Call..10 3.6 Making a Video Call.11 4. Product FAQ...12 5. VoIP FAQ........13 2
1. Overview of V100 Softphone ZyXEL s V100 Softphone is the next generation IP-based video communication software, which allows mobile workers to efficiently communicate by using their laptop or PC. The soft phone helps to enable voice and video communication based on the SIP infrastructure. ZyXEL V100 Softphone will be delivered as a feature of ZyXEL IP-PBX. Hence, the V100 Softphone will be only allowed to register to a ZyXEL IP-PBX. V100 supports auto-provision with ZyXEL IP-PBX, so users don t have complicated setting on V100. Configuration would be set up by auto-provision. 3
2. Setting up the V100 Softphone a. Double click "setup.exe" file b. Click Next to continue Figure 1. Click Next c. Click "I Agree" to agree license agreement Figure 2. Click I Agree 4
d. Select the path you want to install V100 and click "Install" Figure 3. Select the path to install V100 e. Click "Finish" to finish the V100 installation Figure 4. Complete V100 installation 5
f. After successfully installed V100 into your PC, click the V100 icon to execute the soft phone. g. Click the icon in the tool bar to open the Setting menu. Figure 5. Click the Setting icon to open the Setting menu h. Click the SIP tab and you will see the screen as below. Figure 6. Complete V100 installation 6
i. Enter Serial Number that you configured on the X6004 and enter the X6004 IP address to do Auto Provision. j. Enter the Account name to show on the V100 main screen after register to X6004 k. Enter Number that you create on the X6004 l. Enter User Name that you create on the X6004 m. Enter Password that you create on the X6004 n. Enter X6004 IP address (LAN or WAN) on the Domain. In this example, LAN IP address is used. o. Enter X6004 IP address (LAN or WAN) on the SIP Server. In this example, LAN IP address is used. p. Enter X6004 IP address (LAN or WAN) on the Backup Domain. In this example, WAN IP address is used. q. Enter X6004 IP address (LAN or WAN) on the Backup SIP Server. In this example, WAN IP address is used. r. Click Apply to finish the setting 3. V100 Basic Phone Usage 3.1 Making a call When making a call, enter the number and press the Dial button. See figure 7. Figure 7. Enter the phone number and press the Dial button 7
3.2 Receiving a call When the phone rings, press the Dial button to receive the call. When the call is established, the status bar displays Talking. Figure 8. Press the Dial button to answer the incoming call 3.3 Ending a call When you want to end a call, press the End button. Figure 9. Press the End button to finish the call 8
3.4 Placing a call on Hold When you place a call on hold, press the Hold/Resume button. ZyXEL V100 Support Notes Figure 10. Press the Hold/Resume button to place the call on hold Press the Hold/Resume button again to return to the call. See figure 11. Figure 11. Press the Hold/Resume button again to resume the call 9
3.5 Transferring a call During the ongoing call, click the Transfer button or dial *96, which is a feature code assigned in the IPPBX and the next available line will activate. Then dial the extension to which you want to transfer the call. After the extension is ringing, press the End button on V100 and answer the call at the extension, then the call will be transferred to the extension you dialed successfully. Figure 12. Click the Transfer button and 1001 to transfer a call to another extension 1001. Figure 13. Dial *96 and 1001 to transfer a call to another extension 1001 10
3.6 Making a video call The V100 uses the list of cameras detected by your computer. Before you begin to make a video call, please ensure the camera is correctly installed and recognized by your operating system. a. Click the icon in the tool bar to open the video window. b. Enter the number you want to call and press the Dial button. c. Click Yes. Then Ring will shows in the display screen. When the called party answers, Talking will show in the display screen and the video call is established. d. Click Send/Stop button to send/stop video stream. Figure 14. Click the icon in the tool bar to open the video window 11
4. Product FAQ What is the V100 Softphone? The V100 Softphone is an IP-based video communication software, which allows mobile workers to efficiently communicate by using their laptop or PC. The softphone helps to enable voice, video and text-based instant messenger communication based on the SIP infrastructure. What audio codec does V100 support? V100 supports the following commonly used codec. G.729a/b voice codec G.711u-law voice codec G.711a-law voice codec G.726 voice codec What video codec does V100 support? MPEG-4 H.263 H.264 What method does V100 support for the NAT traversal? V100 supports Outbound Proxy for the NAT traversal solution. What call features does V100 support? V100 supports Call Waiting, Call Forward (DND/Blind/Busy/No Answer), Call Transfer (Blind Transfer/ Consultant Transfer), Call Hold/Call Retrieve, Call Mute and 3-way conference (Audio). What operating system does V100 support? V100 supports Microsoft Windows 2000, XP and Vista. What is the maximum number of buddies that can be added in the Buddy list of V100? In the current version, you can add up to 30 buddies in the Buddy list. 12
I obtain a sip account from other sip account providers and it can be successfully registered when I use the softphone eyebeam. Why can't this sip account be registered when I use the V100? V100 Softphone will be delivered as a feature of ZyXEL IPPBX. Hence, V100 will be only allowed to register to a ZyXEL IPPBX. 5. VoIP FAQ What is Voice over IP? Voice over IP is an emerging technology based on open standards of IEEE, fundamentally the Internet Protocol, which allows voice data to travel across the Internet. There are many methods using this technology, the most common and well known are SIP, and H.323. How does Voice over IP 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. Why use VoIP? Traditionally telephony carrier use circuit switching for carrying voice traffic. As circuit switching is designed to carry voice and it does it very well. Then 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 cannot 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. What is the relationship between codec and VoIP? In order to transfer voice (analog signal) over IP it first needs 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 its advantage and disadvantage. 13
What advantage does Voice over IP 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) cannot. What is the difference between H.323 and SIP? H.323 and SIP are proposed by different group Session Initiation Protocol (SIP) is a standard introduced by the Internet Engineering Task Force in 1999 to carry voice over IP. Since it was created by the IETF, it approaches voice and multimedia from the Internet, or IP, perspective of view. Whereas H.323 emerged around 1996, and as an International Telecommunication Union standard it was designed from a telecommunications perspective. Both standards have the same objective - to enable voice and multimedia convergence with IP protocols. Can H.323 and SIP interoperate with one another? In interoperability between the two, the industry is making slow but sure progress. Interoperability must first happen between vendor implementations of the same protocol (SIP-to-SIP and H.323-to-H.323) and then between protocols. Currently in order for SIP client to talk to H.323 client, the ITSP must have a trunking gateway acting as a translator between the two protocols. Without the trunking gateway, the two protocols are not able to communicate to each other. What is voice quality? Voice quality is how well a person can hear the voice on the opposite end. 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. What is codec? Codec is an algorithm which converts analog signal into digital signal and vice versa. There are three main types of waveform codec, source codec, and hybrid codec. Each consume different amount of bandwidth and provide different voice quality level. 14
What is the relation of codec and VoIP? VoIP sends voice information in digital form in discrete packets over digital network and this digital network is public network, thus there may be 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 bandwidthwise aspect the smaller amount of bandwidth used the better. But in voice aspect the higher quality the better. 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. What do I need in order to use SIP? The minimum required to use VoIP is as follow. 1. A high-speed Internet connection. This can be a cable modem, or a high-speed network services such as ISDN, DSL or a T-1 link. The need of the bandwidth required will depend on the amount of telephone traffic will be in your network. 2. A PC with VoIP software installed or a hardware VoIP box such as ATA or device like V500 IP Phone or VoIP station router. 3. An account with a VoIP provider such as an ITSP. The account can be configured to recognize your calls automatically, or you can require the users to enter their unique account numbers issued. Unable to register with the ZyXEL IPPBX. If you are unable to register with ZyXEL IPPBX, 1. Make sure the Internet is reachable and the IPPBX is reachable. 2. Make sure the SIP account and password are correct. 3. Check if there is NAT router before it. If there is a NAT router before it, check the NAT traversal method is configured. V100 support Outbound Proxy method for the NAT traversal solution. 15
I can register but cannot establish a call? If you can register to server but cannot make a call very likely there is NAT router or firewall before it which is blocking it. If you have an NAT router before it, please make sure the NAT traversal method is enable. If you have a firewall before it, please check with the firewall manager. Make sure the SIP protocol is allow to pass-through firewall, and the range of RTP port is allowed through firewall. I can make/receive a call but the voice only goes one way not both ways? If the call can be established but the voice only goes one way, it is very likely that there is NAT router or firewall before it. Please see NAT/firewall related question above. 16