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 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 combined 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 Hanceville.
Manufacturers of SIP equipment are quickly crafting innovative hardware and software to get the upper hand in 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 plan in your company, you should take into account the appropriate configuration of your corporate firewall to accept SIP. Many of the common firewalls already deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are typically closed on firewalls. Second, SIP clients within 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 provider is able to handle 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 problems. By including a SIP proxy and registrar for managing the firewall, it is feasible to handle complex SIP situations for secure and confidential communications.
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