RTP stands for Real-time Transport Protocol and it’s the protocol that allows the actual audio of a VoIP phone call to be heard at either end. RTP can pretty much communicate over any port (Lets call that a door on your network) on the Internet, but most common is a port between 10,000 and 40,000. You need one port for each leg of a VoIP phone call, so one for you and one for the person you are talking with.
The exact specification for the core of RTP can be found on the Internet Engineering Taskforce’s website.
The toughest part of RTP is simply agreeing which door to have a conversation over. This is all negotiated as part of a SIP conversation at the “Invite” stages, so although tough (Simply as its not known or agreed until this stage) SIP takes care of all of that for you.
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