Hi Shamus,

I'm glad you solved your own problem.  Yes, the dynamic, connectionless nature 
of RTP is not simple math as you had hoped.

I've give you one example, since RTP is over UDP/IP there is no 'connection 
state' as TCP/IP has, so when a SIP call terminates, the firewall maintains the 
RTP 'conntrack' for 3 minutes and the network stack for some period of time as 
well, since there is no way to signal that UDP is 'done', it simply times out.  
Given that, it is easy to see how a small RTP range could fill-up and appear 
'busy' to asterisk.

I'm sure others here could explain better than I, nevertheless you need extra 
headroom for your RTP range, greater than simple math would suggest.  Given 
that the default rtp.conf range is 10000, you can make it much, much smaller, 
but I'd probably set it at least 128 (rtpend-rtpstart).  I use 256.

Lonnie



On Feb 28, 2012, at 7:04 PM, Shamus Rask wrote:

> I'm currently running AstLinux 1.0.2 (thank you team, flawless upgrade!) with 
> Asterisk 1.8.x. My SIP provider allows for up to 5 concurrent calls; I had 
> limited my port range in both rtp.conf and my firewall to 10000 - 10020. 
> However, I recently noticed that one of my 3 SIP extensions stopped ringing.
> 
> Looking at the CLI, I see the following error:
>> [Feb 28 19:46:54] WARNING[6396]: app_dial.c:2218 dial_exec_full: Unable to 
>> create channel of type 'SIP' (cause 20 - Unknown)
>> [Feb 28 19:46:54] ERROR[6396]: res_rtp_asterisk.c:511 ast_rtp_new: Oh 
>> dear... we couldn't allocate a port for RTP instance '0x8885d98'
> When I doubled the number of RTP ports available in rtp.conf, all extensions 
> rang as expected.
> 
> I'm not implying this is an AstLinux issue, but am hoping this community can 
> help me better understand how many RTP ports could be consumed? My goal is to 
> limit the number of open ports on my firewall.
> 
> many thanks,
>    Shamus



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