Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person real time traffic over the Web. Its technical specifications come from the SIP working group of the Internet Engineering Task Force. SIP provides 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. Future enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Laurel.
Producers of SIP hardware are rapidly crafting new products and software to get the upper hand in this new Internet communications method. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before installing a SIP solution in your business, you should take into account the proper settings of your corporate firewall to accept SIP. Many of the typical firewalls already deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are normally blocked on firewalls. Second, SIP clients within a firewall can not be accessed using IP addresses since these addresses are local and unique 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 manager 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 possible to handle complex SIP scenarios for secure and confidential communications.
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