Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol for person-to-person voice traffic 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. Future enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in North Branch.
Makers of SIP equipment are quickly designing new products and software to take advantage of this new Internet communications method. 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 setting up a SIP plan in your company, you should take into account the proper configuration 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 transmitted over dynamically allocated UDP ports that are often 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 carrier is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to properly support SIP to get around these technical challenges. By including a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP situations for secure and private communications.
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