SUSE Linux 10 for dummies phần 7
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SUSE Linux 10 for dummies phần 7 165 Chapter 10: E-Mailing, IM-ing, and Using IP Phone in SUSEMaking IP Telephone Calls in SUSE Linux You may have heard about VoIP (Voice over Internet Protocol) — a technol- ogy that converts analog voice signals into digital data packets that can then be sent over the Internet to its destination, where the data packets are con- verted back to analog audio. VoIP, also known as IP telephony, enables voice communications over the Internet, and applications are available to make such IP telephone calls from your SUSE Linux system. As you might guess from my rough description of VoIP, you need a way to get your voice into the computer and listen to the voice signals from the person you are talking to. Your system’s sound card (with a microphone plugged in for voice input) and the speakers attached to the sound card can take care of the audio side of the phone call. If you want to routinely make IP phone calls, you may want to buy a headset with a built-in headphone and microphone. Internet telephones use one of two protocols — H.323 and SIP (Session Initiation Protocol) — for sending voice over IP networks. H.323 is an older protocol that’s supported by Microsoft Netmeeting and GnomeMeeting (avail- able in SUSE Linux’s GNOME desktop). SIP is an Internet standard protocol, and many IP telephony devices and applications support it. For example, KPhone and Linphone — two IP telephony programs included in SUSE Linux — support SIP. For SIP-compliant IP telephony applications, you have to use a SIP URL with the syntax sip:user@hostname or sip:user@IPaddress (where IPaddress is an IP address) to identify the person you are calling. A SIP phone application can call another directly, provided that both applica- tions can reach each other on the IP network. For example, on my home LAN, I can start KPhone on a Linux PC with the IP address 192.168.0.2 and use the URL sip:naba@192.168.0.6 to call a user on another system on the LAN. Similarly, you can call anyone else running a SIP phone application on a system with a public IP address. To call other SIP phones on the Internet and receive incoming calls without having to know everyone’s public IP address (or to do so in cases in which the systems are behind firewalls or NAT routers), you need to register with a registration service such as FreeWorldDialup, or FWD (www.freeworlddialup.com), which is a free service. When you register with FWD, you get a username, password, and a six-digit SIP number. That SIP number is what others use to call you. For example, if your FWD number is NNNNNN, your SIP URL is sip:NNNNNN@fwd.pulver.com. For a list of other public SIP servers and information on how you can register and what the SIP URL looks like, see www.linphone.org/doc/us/servers. html and www.cs.columbia.edu/sip/servers.html. You can also find lots of useful IP telephony links at www.cs.columbia.edu/sip/ and www.voip-info.org/tiki-index.php.166 Part III: Doing Stuff with SUSE Typically many home LANs and PCs are behind a NAT router, so your PC does not have a public IP address. To get over this problem, you can use STUN (Simple Traversal of UDP over NAT), which is a protocol that enables an IP phone to get information from a public STUN server about your NAT and automatically modify the IPaddress and port so that the phone call works through the NAT. If the IP telephony application does not support STUN, you can use what is known as an outbound proxy — another public server that enables IP telephone calls to work over a NAT. FWD provides a STUN server at stun.fwdnet.net:3478 and an outbound proxy server at fwdnat. pulver.com:5082. You need this information to configure IP telephony applications such as KPhone and Linphone. Using KPhone or Linphone You can use the KPhone or Linphone application to make IP phone calls in SUSE Linux. If either of these applications is not installed, you can install each by running the YaST Control Center. Select the Software category and click Software Management. Then search for the application’s name (for example, Linphone), select the package, and click Accept. To make IP phone calls with KPhone or Linphone, here are the general, high- level steps: 1. Connect the microphone to the sound card and test it with the sound recorder ...
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SUSE Linux 10 for dummies phần 7 165 Chapter 10: E-Mailing, IM-ing, and Using IP Phone in SUSEMaking IP Telephone Calls in SUSE Linux You may have heard about VoIP (Voice over Internet Protocol) — a technol- ogy that converts analog voice signals into digital data packets that can then be sent over the Internet to its destination, where the data packets are con- verted back to analog audio. VoIP, also known as IP telephony, enables voice communications over the Internet, and applications are available to make such IP telephone calls from your SUSE Linux system. As you might guess from my rough description of VoIP, you need a way to get your voice into the computer and listen to the voice signals from the person you are talking to. Your system’s sound card (with a microphone plugged in for voice input) and the speakers attached to the sound card can take care of the audio side of the phone call. If you want to routinely make IP phone calls, you may want to buy a headset with a built-in headphone and microphone. Internet telephones use one of two protocols — H.323 and SIP (Session Initiation Protocol) — for sending voice over IP networks. H.323 is an older protocol that’s supported by Microsoft Netmeeting and GnomeMeeting (avail- able in SUSE Linux’s GNOME desktop). SIP is an Internet standard protocol, and many IP telephony devices and applications support it. For example, KPhone and Linphone — two IP telephony programs included in SUSE Linux — support SIP. For SIP-compliant IP telephony applications, you have to use a SIP URL with the syntax sip:user@hostname or sip:user@IPaddress (where IPaddress is an IP address) to identify the person you are calling. A SIP phone application can call another directly, provided that both applica- tions can reach each other on the IP network. For example, on my home LAN, I can start KPhone on a Linux PC with the IP address 192.168.0.2 and use the URL sip:naba@192.168.0.6 to call a user on another system on the LAN. Similarly, you can call anyone else running a SIP phone application on a system with a public IP address. To call other SIP phones on the Internet and receive incoming calls without having to know everyone’s public IP address (or to do so in cases in which the systems are behind firewalls or NAT routers), you need to register with a registration service such as FreeWorldDialup, or FWD (www.freeworlddialup.com), which is a free service. When you register with FWD, you get a username, password, and a six-digit SIP number. That SIP number is what others use to call you. For example, if your FWD number is NNNNNN, your SIP URL is sip:NNNNNN@fwd.pulver.com. For a list of other public SIP servers and information on how you can register and what the SIP URL looks like, see www.linphone.org/doc/us/servers. html and www.cs.columbia.edu/sip/servers.html. You can also find lots of useful IP telephony links at www.cs.columbia.edu/sip/ and www.voip-info.org/tiki-index.php.166 Part III: Doing Stuff with SUSE Typically many home LANs and PCs are behind a NAT router, so your PC does not have a public IP address. To get over this problem, you can use STUN (Simple Traversal of UDP over NAT), which is a protocol that enables an IP phone to get information from a public STUN server about your NAT and automatically modify the IPaddress and port so that the phone call works through the NAT. If the IP telephony application does not support STUN, you can use what is known as an outbound proxy — another public server that enables IP telephone calls to work over a NAT. FWD provides a STUN server at stun.fwdnet.net:3478 and an outbound proxy server at fwdnat. pulver.com:5082. You need this information to configure IP telephony applications such as KPhone and Linphone. Using KPhone or Linphone You can use the KPhone or Linphone application to make IP phone calls in SUSE Linux. If either of these applications is not installed, you can install each by running the YaST Control Center. Select the Software category and click Software Management. Then search for the application’s name (for example, Linphone), select the package, and click Accept. To make IP phone calls with KPhone or Linphone, here are the general, high- level steps: 1. Connect the microphone to the sound card and test it with the sound recorder ...
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