Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol for person-to-person real time data over the Web. Its defining 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 deal with IP telephony in conjunction with video and instant messaging. Forthcoming improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Berlin.
Producers of SIP hardware are quickly developing innovative products and software to take advantage of this new Internet communications scheme. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before setting up a SIP plan in your business, you should take into account the appropriate configuration of your corporate firewall to support SIP. Many of the common firewalls already deployed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are often shut on firewalls. Second, SIP clients inside a firewall can not be reached using IP addresses because these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier 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 get around these technical difficulties. By including a SIP proxy and registrar for managing the firewall, it is feasible to handle complex SIP scenarios for reliable and confidential communications.
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