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Embedded communication can take different shapes and sizes. If used with web browsers, communication relies on WebRTC for media processing with additional signaling. On mobile devices and other native applications, embedded communication can use virtually anything, including open source and commercial media engine stacks, to communicate with an internet protocol PBX.
To connect to an IP PBX or telephony infrastructure, embedded communication needs to be translated into a format or protocol that the IP PBX can understand. In almost all cases today, that means turning to the Session Initiation Protocol (SIP). IP PBX and telephony infrastructures commonly use SIP as the glue that keeps things together. SIP is what allows for connections with an IP PBX infrastructure.
The IP PBX signal is translated to SIP, while the media itself may change its transport protocol and may be transcoded if there's a mismatch in the supported codecs. For example, when you connect WebRTC-based embedded communication with a telephony infrastructure, you will likely transcode from an Opus codec to a voice codec, such as G.711.
One thing to note is that SIP comes in many forms, and interoperating with a specific IP PBX or infrastructure can be somewhat challenging. SIP deployments come in many flavors, and vendors often offer their own implementations with proprietary features. In some cases, these proprietary features are necessary for successful communication in an organization's specific deployment. In that case, one or both ends of the connection will need to be slightly modified with flexible gateways or session border controllers.
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