Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the communications technology for person-to-person voice data over the Web. 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 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 Rumford Center.
Suppliers of SIP hardware are rapidly designing new hardware and software to take advantage of this new Internet communications scheme. SIP telephones, PC client software, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP solution in your organization, you should consider the proper configuration of your corporate firewall to accept SIP. Many of the typical firewalls currently deployed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are normally blocked on firewalls. Second, SIP clients within a firewall can not be reached 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 handle the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to appropriately support SIP to get around these technical challenges. By adding a SIP proxy and registrar for controlling the firewall, it is possible to handle complex SIP scenarios for secure and confidential communications.
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