Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport 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 enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Currant.
Suppliers of SIP components are quickly crafting innovative hardware and software to get the upper hand in this new Internet communications method. SIP telephones, PC client software, 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 company, you should consider the proper settings of your corporate firewall to accept SIP. Many of the typical firewalls currently deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are usually closed on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses since 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 handle 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 managing the firewall, it is feasible to deal with complex SIP situations for secure and confidential communications.
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