[asterisk-users] Question about how Asterisk works with RTP ports

[Jonas Kellens]

Hello,

short question : does Asterisk reserve RTP ports for every IP-phone that 
is being called ?

If for instance an incoming call makes 10 IP-phones ring, does this mean 
that Asterisk preserves 10 x 2 RTP ports for audio ?

I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port 
number for audio ? If this is the case for the 10 IP-phones to which an 
INVITE is send to, this means at least 10 RTP ports are reserved for 
incoming audio, correct ???



Thanks.

Jonas.

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