Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person voice 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 combined 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 Bolivar.
Manufacturers of SIP hardware are quickly developing new hardware and software to take advantage of this new Internet communications scheme. 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 company, you should take into account the appropriate configuration of your corporate firewall to support SIP. Many of the common firewalls already installed in business offices are not designed to support the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are usually blocked on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses since these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband provider or Ethernet fiber provider is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to appropriately support SIP to overcome these technical challenges. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to handle complex SIP situations for reliable and confidential communications.
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