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SIP connections can be made in two primary ways--registered and unregistered. In unregistered mode, a SIP channel is
opened between devices at the time a call is placed. In registered mode, a SIP channel is constantly maintained between a
SIP client (like a studio talkshow system) and a SIP server (like that at an Internet Telephone Provider). Most VoIP users will
only use registered mode, so that’s what we’ll focus on going forward.
The SIP protocol can be used in more than one link in a VoIP chain. The best example would be a purely IP PBX. In this case,
the PBX maintains a SIP channel to an Internet Telephone provider on its WAN port. It also maintains several SIP connections
over its LAN to telephone extensions. Because the protocol used in these links is identical, it provides for a lot of flexibility. For
example, if need be, the telephone extensions could register directly with the provider, bypassing the PBX entirely.
It’s important to understand that the SIP protocol does not carry any actual voice between devices-- it simply instructs devices
to create separate RTP sessions in each direction. RTP streams are created and destroyed based on commands contained in
SIP messages when calls are made or received.
Summary of Contents for VH2
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