Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology 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 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 handle 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 companies in White River Junction.
Makers of SIP equipment are quickly developing innovative hardware and software to take advantage of this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to implementing a SIP solution in your organization, you should take into account the proper settings of your corporate firewall to support SIP. Many of the typical firewalls currently deployed in business offices are not designed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are usually closed 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 carrier or Ethernet fiber carrier is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to properly support SIP to get around these technical challenges. By including a SIP proxy and registrar for managing the firewall, it is possible to deal with complicated SIP scenarios for secure and confidential communications.
