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

[Jonas Kellens]

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.