Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport technology for person-to-person real time data over the Internet. 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 expanded to deal with IP telephony in conjunction with video and instant messaging. Forthcoming enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Basin.
Suppliers of SIP equipment are rapidly creating new products and software to take advantage of this new Internet communications method. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before implementing a SIP solution in your organization, you should take into account the proper configuration of your corporate firewall to accept SIP. Many of the common firewalls currently installed in business offices are not designed to support the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are typically blocked 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 broadband provider or Ethernet fiber service is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to correctly support SIP to overcome these technical challenges. By adding a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP scenarios for secure and confidential communications.
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