When an Avaya SIP Telephone (AST) makes a call to another AST on the same Avaya Aura® Communication Manager (CM), when is a SIP trunk used?

Question

Options

• Aup to the point where calls are shuffled to establish a direct media path
• Bjust during call establishment and clear down only
• Cfor the entire duration of the call
• Donly when the media path is established

• B
  5% (2)
• C
  92% (36)
• D
  3% (1)

Explanation

When ASTs communicate on the same CM, the SIP trunk supports an active SIP dialog that must remain open from the moment the call is established until it is torn down with a BYE message - this is fundamental to how the SIP protocol works, making C correct. Unlike H.323 or TDM calls, SIP does not release the signaling path mid-call; the dialog persists for the entire session to handle in-call features (hold, transfer, re-INVITE for codec renegotiation, etc.).

Why the distractors are wrong:

• A confuses the concept of media shuffling (where RTP can be redirected to a direct endpoint-to-endpoint path) with the signaling trunk - even after media is shuffled, the SIP signaling trunk stays active for the full call.
• B describes a misconception borrowed from circuit-switched thinking, where a "control channel" is only needed at setup/teardown; SIP dialogs don't work that way.
• D inverts the reality - the SIP trunk is used before and throughout media flow, not only at the moment media is established.

Memory tip: Think of a SIP trunk as a "phone call lease" - you hold it for the entire duration you're on the call, not just when you pick up and hang up. SIP = Session (ongoing), not Signal-only.

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