Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission 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 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 handle IP telephony in conjunction with video and instant messaging. Future improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Forest Ranch.
Makers of SIP hardware are rapidly creating innovative products and software to take advantage of this new Internet communications method. SIP telephones, PC client software, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before setting up a SIP plan in your business, you should think about the appropriate settings of your corporate firewall to accept SIP. Many of the typical firewalls already installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are normally closed on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses because these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband provider or Ethernet fiber provider is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to correctly support SIP to overcome these technical issues. By including a SIP proxy and registrar for controlling the firewall, it is feasible to handle complex SIP situations for secure and private communications.
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