The Session Initiation Protocol (SIP) is basically responsible for establishing communication between two devices. More technically speaking, it is an open signaling protocol used to establish, modify and end communication sessions through an IP network together with two RTP / RTCP protocols and the SDP (Session Description Protocol) protocol. These sessions can be as simple as voice communication between two points, or more complex as in the case of a multi-party web conference with voice, video and document sharing.
SIP has a significant resemblance to the HTTP protocol, which makes it easier to understand and solve problems. SIP is also independent of the medium used. It can work for voice, video or instant messaging. The messages are text-based and the request-response mechanism makes the resolution of errors very easy.
What is a SIP trunk?
In traditional telephony, calls are made via the traditional telephone network (RTC), a dedicated network in analog or digital form. As SIP is an IP protocol, it operates on the same network as data traveling through the Internet. This convergence of voice and data means that a SIP trunk is more about the bandwidth than a cable network or a physical circuit. These are some of the benefits of SIP Trunks :
Convergence of voice and data Reduction of equipment, saving money, space and energy
Flexible costs thanks to a strong competitive offer Improved reliability and redundancy
Traditionally, CTI (Computer Telephony Integration) technology, which is the integration between data and telephony, has been mainly focused on controlling physical devices (for example, the telephone), without taking into account the user of the same. However, with the explosion of communication interfaces, it is likely that the user will use many devices or SIP clients during the day. For example, a business executive will surely use a desktop IP phone , a SIP client on his mobile phone , a softphone on his computer and an instant messaging client .
An example of this CTI integration is when we receive a call, we obtain information about the call identifier and compare it with our records within productivity applications such as a CRM; with it we know who calls us, the contact data and their call history and even previous sales or interactions. These functions allow communications to be more effective.
With SIP, the user’s communication activities, including presence data (“absent”, “do not disturb”, etc.) are handled as a whole. This user-centric model is a revolution compared to a device-centric model where each entity is treated separately and without a particular connection with its owner.
Just as you would never connect a computer to the Internet without configuring the appropriate security tools (Firewall, antivirus), a SIP user (VoIP) also needs protection against malicious activity. SIP has many mechanisms to ensure its communication.
For example, a SIP call can be encrypted through TLS and SRTP; and the best performing VoIP communications solutions can block SIP attacks by blacklisting some malicious IP addresses. Finally, some components, such as a Session Border Controller (SBC), can be implemented as protection for remote entities following the same principle as a Network Firewall but in this case for SIP.
This SIP Protocol guide shows that with the right information, it is easy to understand what is the communication technology of the present and certainly of the future. Sip Trunking For Testing The sooner you consider migrating your old PSTN phone lines, the sooner you can begin to enjoy the benefits of a low-cost, reliable technology, suitable for all businesses, regardless of their size.