SIP (Session Initiation Protocol) and WebRTC have a unique relationship with one another. While both are infrastructures developed to support real-time communication and collaboration over the Internet, each method varies greatly in operation and capability. Despite a seemingly similar appearance, these methods are not competitors as much as they are siblings. WebRTC is not a newer form of SIP. Instead, WebRTC, like SIP, is a VoIP technology that expands on and integrates SIP functionality. That being said, the two infrastructures embrace a symbiosis in which one compliments the other.

Over the past decade, SIP has become the predominant protocol used to set up real-time media sessions between groups of users. Thusly, there are now a number of SIP providers available to users.  In doing so, the protocol is able to set up simple telephone calls, video, and audio for multicast meetings, or instant messages (conferencing). Inversely, WebRTC is a communications technology that looks to add real-time media—i.e. audio, video, file transfer—to every web browser. In doing so, computers, phones, tablets, and other devices no longer need to install softphones. Instead, every device with a web browser will have real-time communications capabilities. While WebRTC looks poised for various applications that range from online gaming to business, there is an issue of connectivity—i.e. a protocol is necessary. This is where SIP comes in.

SIP’s primary functionality lies in setting up connections between groups of users/participants. While SIP devices can communicate directly with one another, they often implore other intermediary systems (SIP proxy) and other protocols to connect SIP servers to SIP endpoints. Inversely, WebRTC only sets up and describes the media and its capabilities; therefore, a method of exchange is still necessary for a session to be established. While WebRTC will work fine for users that want to enhance an existing service with real-time audio and video, a protocol is needed to move past this function and communicate with others. In regards to the necessary protocol, WebRTC will need one that reaches out and sets up a session. This is the exact function of SIP.

The main method of functionality for SIPs is as follows: it operates on a session, finds the other party/parties, sets up the session, manages the session, and ends the session. This is exactly what WebRTC needs; however, is this dependence one-sided? Do SIPs need WebRTC? The answer is no. SIPs can use multimedia systems on a computer without a browser. For example, users can use SIP with a VoIP provider and softphone software. However, from a user’s point of view, WebRTC makes SIPs easier to use. For example, WebRTC uses the device’s browser—which is already installed on the computer; therefore, it’s much easier to use. As stated above, you don’t have to install any additional software (softphones); therefore, you don’t have to learn how to use new applications/devices. Instead, you can use your browser—which you are already familiar with.

SIP and WebRTC are both methods of VoIP as they both stand for real-time communications and look to send voice (and video) over an IP network (using the same standards/codecs). Yet, despite these similarities, these technologies better represent two halves of one whole. WebRTC does not need to use SIP—it can function with another protocol, or without one altogether. Additionally, SIP does not need WebRTC—it can function alone or by utilizing other protocols (for example Real-time Transport Protocol), SIP proxy servers, registrars (requests and places information to correlating locations), redirect servers, session border controllers, and/or gateways to send voice data between phones after setting up calls. Yet despite their singular functionalities, both technologies benefit from the other’s inclusion.