[Asterisk-Users] Narrowing RTP port range

[Leo Ann Boon]

Rich Adamson wrote:

>>I'm trying to lock down my asterisk install as much as possible and I
>>keep reading about people saying 'you can narrow the range of ports in
>>rtp.con' (by default it's from 10000 to 20000 I think).
>>
>>My question is this - how much can I narrow it down?  Can I narrow it to
>>10 ports, or can the ports not be reused for additional conversations?
>>
>>I guess what I'm asking is - does the number of ports in the range have
>>anything to do with the number of simultaneous connections or anything
>>like that?
>>    
>>
>
>Yes, you can narrow it down. One port will be required for each
>leg of a call. So if sip/123 called sip/345, that's two ports. I'd
>suspect that some ports are used for other purposes besides just a
>conversation, so adding more is certainly more in your best interest
>then cutting them short.
>  
>
You are right, each RTP stream requires 2 ports. Each RTP port has an
associated RTCP port using  the next higher odd number port. I.E, if the
first RTP port assigned is 10000, the next will be 10002, not 10001 as
that would be the RTCP port for 10000. In practice, it's possible to use
odd number ports for RTP (works on my old ATA-186).  But it's really bad
idea, because there're phones that are hardcoded to look for RTCP in odd
number ports.