Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol for person-to-person voice data over the Web. Its defining 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 handle IP telephony in conjunction with video and instant messaging. Forthcoming enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Mayo.
Makers of SIP hardware are rapidly designing new hardware and software to get the upper hand in this new Internet communications method. SIP telephones, PC client software, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Before setting up a SIP plan in your company, you should think about the appropriate configuration of your corporate firewall to accept SIP. Many of the common firewalls already installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are usually shut on firewalls. Second, SIP clients inside 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 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 appropriately support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complex SIP situations for secure and private communications.
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