Why You Need SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person voice data over the Web. Its defining 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 expanded to deal with IP telephony in conjunction with video and instant messaging. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Des Peres.
Manufacturers of SIP components are quickly crafting new hardware and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Prior to implementing a SIP solution in your organization, 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 inside 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 broadband provider or Ethernet fiber service 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 get around these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complex SIP situations for secure and confidential communications.
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