Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications protocol for person-to-person real time traffic over the Internet. 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 enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Cave City.
Producers of SIP equipment are rapidly creating new products and software to take advantage of this new Internet communications method. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to implementing a SIP solution in your business, you should think about the appropriate configuration of your corporate firewall to accept SIP. Many of the common firewalls already deployed 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 shut on firewalls. Second, SIP clients within a firewall can not be accessed using IP addresses since these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line carrier or Ethernet fiber service is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to correctly support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to handle complicated SIP situations for secure and confidential communications.
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