On 10/29/2013 05:14 PM, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?
It uses 2 ports per channel under normal circumstances, 1 for RTP and
1 for RTCP.
If for instance an incoming call makes 10 IP-phones ring, does this mean
that Asterisk preserves 10 x 2 RTP ports for audio ?
Yes.
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 ???
Yes.
So if I understand correct, you don't need to look at the amount of
concurrent calls to calculate the RTP range in rtp.conf, you need to
look at the amount of INVITES that are being send at one moment ?
Kind regards,
Jonas.
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