Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol for person-to-person voice traffic over the Web. Its technical specifications come from the SIP working group of the Internet Engineering Task Force. SIP provides access to the public switched telephone network for voice data at 3 kHz bandwidth and common number dialing using Voice over IP (VoIP). It can also be extended to deal with IP telephony in conjunction with video and instant messaging. Future enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Kingsville.
Manufacturers of SIP components are rapidly crafting innovative products and software to take advantage of this new Internet communications method. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before setting up a SIP solution in your company, you should think about the proper settings of your corporate firewall to support SIP. Many of the common firewalls already installed in business offices are not designed to support the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are often shut on firewalls. Second, SIP clients inside a firewall can not be reached using IP addresses since these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Ethernet fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to appropriately support SIP to overcome these technical problems. By including a SIP proxy and registrar for controlling the firewall, it is possible to deal with complex SIP situations for secure and confidential communications.
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