Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission protocol for person-to-person voice data 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. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Santan.
Producers of SIP components are rapidly designing innovative products 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 obtainable from companies such as Ingate Systems and Cisco.
Before installing a SIP plan in your company, you should think about the appropriate configuration of your corporate firewall to accept SIP. Many of the typical firewalls already installed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are often shut on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses because these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line provider or Metro fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to properly support SIP to get around these technical issues. By adding a SIP proxy and registrar for managing the firewall, it is possible to deal with complex SIP situations for secure and confidential communications.
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