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.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users