In the wake of the COVID-19 pandemic, the shift to work-from-anywhere models for businesses is moving towards permanence. A recent PwC survey among 699 CEOs revealed that nearly 80 percent agreed that remote collaboration will be around for a long time. In another survey conducted by FlexJobs, 65 percent of employees reported that they plan to do remote work full-time after the pandemic, while 30 percent prefer a hybrid setup.
This level of flexibility could not have been achieved if it weren’t for significant advances in tech communication, including developments in session initiation protocol.
What is Session Initiation Protocol?
Also known as SIP, a session initiation protocol is a signaling protocol that regulates a session between two or more devices or people on an IP network. It can initiate, then modify, and eventually terminate the session, which also involves the exchange of data.
Some may confuse SIP with voice over internet protocol (VoIP) and integrated services digital network (ISDN). Although they may function similarly, there are pronounced differences.
One of the biggest differences between SIP and VoIP is the kind of media the former can transmit. VoIP is often limited to phone calls and instant messages while SIP can support video conferencing, internet telephony, and other unified communication options.
Participants can conduct two-way calls or cast multiple sessions simultaneously even without the session’s initiator. SIP has broader uses and is one of the protocols that govern VoIP.
On the other hand, ISDN circuits require the physical installation of equipment in the premises and are highly dependent on the channel blockers used by the client. SIP, therefore, is a more flexible option since it doesn’t demand any physical infrastructure, and data doesn’t need to pass through copper lines. It is also more flexible and scalable since users can add or remove virtual phone lines, making it more cost-effective for businesses that want greater mobility.
Also read: Maximizing the Benefits of UCaaS
SIP Protocol Applications
There are at least two popular applications of SIP: trunking and calling.
SIP trunking refers to the virtual connection between a telephone service provider that can support a unified and SIP-based communication setup and an IP PBX, the virtual or cloud version of the traditional private exchange branch system.
Both are connected over the data network connections used by the business, which may include a wide area network (WAN) and even Wi-Fi. The most important thing to note is that the bandwidth allocation and speed of the internet are enough to meet the needs of the enterprise.
With this setup, two or more parties can then initiate or participate in SIP calling, which is the process of delivering voice calls over a SIP channel or SIP trunk. A channel is a connection that can support both outgoing and incoming calls, and a trunk can have multiple channels.
SIP trunking also helps bypass one of the typical components in a traditional phone system called the primary rate interface line that links the PBX and the public switched telephone network (PSTN). By doing this, an enterprise can still take advantage of the usual call management options like call forwarding without using any analog phone.
There are also different SIP protocol applications depending on the available hardware or software, the type of communication users prefer, budget, and communication policies in an organization. One of the simplest and most popular is the cloud-to-cloud phone system.
How Session Initiation Protocol Works
The basic setup of SIP includes an on-premise or virtual PBX, which is then linked to a SIP trunk, which can then deliver calls and other forms of media via the internet. All of these will also need endpoints to transmit data. One of the most common is the cloud-to-cloud phone system, which one can get on platforms like Skype (users can create their own Skype number).
However, because SIP can support multiple channels, it is also possible to connect a cloud phone system to a non-cloud version, as well as a virtual PBX to a PTSN, which means that someone can actually make an internet call, for example, outside the enterprise network.
To deliver data, such as voice, it needs encoding. Audio signals would have to be converted into codecs, which depends on the allocation or availability of the bandwidth.
For the clearest, best-quality voice data, the ideal codec is G7.11 since it won’t compress it. However, in places where the internet connection is poor or bandwidth is limited, compression may be necessary. Thus, the choice codec is G.729.
SIP isn’t a perfect system. One of the possible SIP protocol vulnerabilities is a malformed attack, whereby one or more participants send a nonstandard data or message usually with the intention of making the system unstable. This will eventually lead to a denial of service (DoS).
However, given the improved cloud security, scalability, and cost-effectiveness of SIP, it still makes a good addition to any business, particularly one that needs to be more mobile.