Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission technology for person-to-person voice data over the Internet. 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 combined 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 Bremen.
Manufacturers of SIP components are quickly crafting new products and software to take advantage of this new Internet communications scheme. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Before setting up a SIP solution in your organization, you should take into account the proper configuration of your corporate firewall to support SIP. Many of the common firewalls currently installed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are typically blocked on firewalls. Second, SIP clients inside 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 carrier or Ethernet fiber service is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to appropriately support SIP to get around these technical issues. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complicated SIP scenarios for reliable and confidential communications.
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