asterisk users - Feb 2013 - RTP timeout if the asterisk box behind NAT

Dears;

I am facing a problem in disconnecting the calls, it is related to the
rtptimeout (disconnecting if there is no RTP packets from both sides).

My Asterisk Box is behind NAT but there is a static real IP address at the ADSL
router. We call from the Mobile to the PSTN analogue numbers which are connected
to Asterisk Analogue card (the telephone lines are analoge), and then we dial
the overseas number, so the asterisk is sending the call to a VoIP service
provider which will route the call to the destination. Sometime the destination
is connected while ringing !! And this is a problem from the SIP service
provider route, then we hangup our mobile (as no one answering our call) but
asterisk is not detecting the hangup (it is because the telephone lines are
analoge and this problem is common in analoge lines that some hangup are not
detected). In that case, the call will stay open and charging and this is a
wrong.

This problem was not appearing when Asterisk machine was having static real IP
address because I was enabling the rtptimeout paramters. But now as the asterisk
box IP address is private and it is behind NATing then it is appearing even I
enabled the (rtptimeout=50 and rtpholdtimeout=120).

What should I do?

Regards
Bilal

On 3/2/13 11:38 pm, bilal ghayyad wrote:
> What should I do?
Given that you said:
> This problem was not appearing when Asterisk machine was having static real
IP address because I was enabling the rtptimeout paramters.
I do believe the solution is simple: put it back on a public IP.

For what it's worth, we have dozens of clients with boxes on RFC1918 IPs 
and we don't see this issue, so I wonder if it's something
'special'
your NAT router's doing to mess up RTP traffic. It's probably worth 
trying a different router (ideally different make/model) and see if 
that's any better. And it's always worth disabling any SIP ALG present 
in the router - they seem to do nothing but break things.

(as a random aside, has anyone *ever* come across a scenario where a SIP 
ALG in a consumer router has actually helped?)

Kind regards,

Chris
-- 
This email is made from 100% recycled electrons

On Sun, 2013-02-03 at 15:38 -0800, bilal ghayyad wrote:
> Dears;
> 
> I am facing a problem in disconnecting the calls, it is related to the
rtptimeout (disconnecting if there is no RTP packets from both sides).
> 
> My Asterisk Box is behind NAT but there is a static real IP address at the
ADSL router. We call from the Mobile to the PSTN analogue numbers which are
connected to Asterisk Analogue card (the telephone lines are analoge), and then
we dial the overseas number, so the asterisk is sending the call to a VoIP
service provider which will route the call to the destination. Sometime the
destination is connected while ringing !! And this is a problem from the SIP
service provider route, then we hangup our mobile (as no one answering our call)
but asterisk is not detecting the hangup (it is because the telephone lines are
analoge and this problem is common in analoge lines that some hangup are not
detected). In that case, the call will stay open and charging and this is a
wrong.
> 
> This problem was not appearing when Asterisk machine was having static real
IP address because I was enabling the rtptimeout paramters. But now as the
asterisk box IP address is private and it is behind NATing then it is appearing
even I enabled the (rtptimeout=50 and rtpholdtimeout=120).
> 
> What should I do?
> 
My advice is to first try to fix your pstn hangup detection problem.
Relying on rtptimeout assumes that the voip side has hanged up and the
voip provider has also terminated the call and no rtp is coming.
Which means that if your pstn caller terminates the call and the voip
side does not (for any reason) you will still be charging the pstn
caller.

To see why rtptimeout does not work get a wireshark capture and see if
there is still traffic going on 
-- 
Stelios S. Koroneos