Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport technology for person-to-person real time traffic over the Web. Its defining specifications come from the SIP working group of the Internet Engineering Task Force. SIP gives 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. Forthcoming improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Port Isabel.
Producers of SIP components are quickly developing new products and software to take advantage of this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Prior to installing a SIP solution in your company, you should think about the appropriate settings of your corporate firewall to accept SIP. Many of the typical firewalls currently installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are typically shut on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses because these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Ethernet fiber provider is able to handle the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to correctly support SIP to get around these technical challenges. By adding a SIP proxy and registrar for controlling the firewall, it is possible to handle complex SIP situations for secure and private communications.
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