Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transport protocol for person-to-person voice 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 combined with video and instant messaging. Future enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Brooklyn.
Makers of SIP components are quickly crafting new hardware and software to take advantage of this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before implementing a SIP plan in your organization, you should take into account the proper settings of your corporate firewall to accept SIP. Many of the typical firewalls already installed 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 inside a firewall can not be reached using IP addresses because these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Ethernet fiber carrier 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 difficulties. By adding 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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