Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications protocol for person-to-person real time data over the Internet. Its technical 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 expanded 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 businesses in Lancaster.
Makers of SIP components are rapidly crafting innovative hardware 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 implementing a SIP plan in your organization, you should consider the appropriate settings of your corporate firewall to accept SIP. Many of the common firewalls already installed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are often closed 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 carrier or Ethernet fiber service is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to appropriately support SIP to overcome these technical issues. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complicated SIP scenarios for secure and confidential communications.
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