![\"\"](\"https:\/\/www.rincon.co.in\/site\/wp-content\/uploads\/2018\/10\/SIP-1024x684.jpg\")
<\/p>\n<\/p>\n
Session Initiation Protocol (SIP) trunking is a method of taking an organization\u2019s complex phone infrastructure and connecting it to the outside world via the internet and VoIP technology rather than physical phone lines (either analog or digital).<\/p>\n
\u201cTrunk\u201d is an old telephony term referring to a location\u2019s connection to the wider grid. \u201cSIP\u201d refers to the software protocol used to begin and end communication connections between devices.<\/p>\n
That said, SIP trunks can also carry video and messaging data \u2013 so think of them as the general communication bridge to the outside world. They\u2019re a common fixture of modern Unified Communications systems.<\/p>\n
A SIP trunk is essentially a more correct way to say \u201cSIP phone line\u201d or \u201cbusiness VoIP line,\u201d although it can do more than those terms imply.<\/p>\n
SIP stands for \u201cSession Initiation Protocol.\u201d SIP is a set of software protocols for initiating and ending connections between devices to send multimedia data (typically via RTP) \u2013 it\u2019s key to much of modern organization-level VoIP data transfer, though it also has other uses.<\/p>\n
Alternative data trunking systems exist that work similarly, but use a different (typically proprietary) protocol, so they\u2019re technically not \u201cSIP\u201d trunks.<\/p>\n
\u201cSIP calling\u201d is a slang term referring to VoIP (Voice over IP) at the organizational level: the process of converting voice audio into data packets and sending them over the same data connections an organization uses for internet access. The system streams them to receiving VoIP hardware, software, or a PSTN gateway (which uses the data to make a traditional phone call closer to the receiving user, a local rather than long distance connection).<\/p>\n
While the two terms are often used interchangeably in casual conversation, they aren\u2019t the same thing. VoIP refers (broadly) to the systems for carrying audio over the internet. SIP is the most popular software protocol used for initiating and ending conversations using that audio. SIP trunking is the term used for connecting office phone systems (using SIP, which not all VoIP systems do) to the internet for VoIP purposes.<\/p>\n
The biggest and best reason organizations use SIP trunks in the short term is cost savings. SIP phone connections tend to be far cheaper (up to 60-80% less) than analog (PSTN) service, so switching can save your business a lot of cash.<\/p>\n
Another good reason is future-proofing. Right now when you make a VoIP call, there\u2019s a good chance that it passes over traditional phone lines at some point in its journey to the recipient. However, that won\u2019t always be the case \u2013 analog phone lines are being phased out. In a few years the analog network won\u2019t be there to connect to, so you might as well switch sooner and start saving money now.<\/p>\n
This can vary from system to system, but a good estimate of an active SIP call\u2019s bandwidth usage is 64k\/s. Because that number is pretty low, the more likely limitation on the number of concurrent calls on a sip trunk is the number of channel licenses the organization has.<\/p>\n
While people sometimes ask \u201chow many SIP trunks do I need?\u201d the real question is \u201chow many SIP channels do I need?\u201d<\/p>\n
Each SIP trunk connects one location to the outside world, while the number of SIP channels within that trunk determine the number of concurrent phone calls that can be active at a time. You typically need far fewer channels than you have extensions, as most extensions won\u2019t be in use at the same time.<\/p>\n
The great thing about SIP channels is that increasing or decreasing them typically doesn\u2019t require corresponding hardware changes. While you\u2019d have to add actual cables to increase an organization\u2019s phone lines, channels are more of a licensing and bandwidth concern.<\/p>\n
More properly called an \u201cIP PBX\u201d, these replace traditional analog PBXs with a more compact & powerful digital alternative. They facilitate the connection of calls to and from your company\u2019s private phone network through a SIP trunk (possibly with a Session Border Controller in between) over the internet, to phones around the world.<\/p>\n
Taking your phone and fax systems into the modern era with VoIP and (its fax equivalent) FoIP can improve efficiency and drop costs, but you can take things even further. Integrating CX-E, XMedius\u2019s premier solution, leverages your existing hardware to deliver a host of powerful applications, such as automated attendants with voice recognition, Interactive Voice Response (IVR) programs, informal call center functionality, unified messaging and a lot more.<\/p>\n
The original article can be found here.<\/a><\/p>\n