dont forget that sip and rtp do not nessecarily go through the same hosts on the same call... how can you monitor rtp on a host if the rtp never goes there Ken Sent from my iPad On Nov 16, 2012, at 3:56 PM, Yiftach Golan <yiftah@choochee.com> wrote:
This is where I am getting a bit confused, if the 200OK arrived to the other side and we checked that the RTP exists (with mod_sofia option) we will not get to hours of calls (unless the other side did not hanged up) In any case there is a good chance that the BYE is getting lost, so the danger exist even without the ACK
I am guessing that the designers of the SIP protocol came up with the ACK because there is a potential for open dialog that is not bound with time and they therefore wanted to know that the other side actually ACKs the request, but since ACK is tied to INVITE only (AFAIK) and INVITE always tied with RTP (at least in most normal SIP implementations) I'm not sure that this ACK is that needed but again as I said it is more of philosophical debate, maybe a potential request in the new RFC
On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice@freeswitch.org> wrote: In this case masking the issue can lead to massive bills... Imaging paying by the minute... All of a sudden you are now leaving 2 minute calls up for hours on end... And continuing to get billed for them... Or you are not continuing to bill a customer for them... And now you have unexpected HUGE bills coming in... Masking it is far worse then just fixing it...
If we mask this one issue, we might as well mask memory leaks, or passwords that don’t work, etc... Sure sometimes we might have to mask an issue for production to work in the short term, but that is never the correct answer fix a problem
On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah@choochee.com> wrote:
While I agree on the details I disagree on the solution Sometimes masking the problems can be a good solution but I guess it is a philosophical debate
On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice@freeswitch.org> wrote: That leaves to big a risk of open sessions and only masks the true issue which is a problem with FS getting the ACK back...
Theres a reason FS is not getting the ACK, and FS will make several attempts to get an ack by retransmitting the 200 OK several times before that timeout occurs.
The real fix here is to fix the underlying cause, not masking it....
On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah@choochee.com <http://yiftah@choochee.com> > wrote:
I know that it is kind out of the what RFC3261 instructs, but did anyone think on giving the option in configuration not to hang up calls in case of an ACK does not arrive? I know that it has the risk of open sessions but there some other ways to handle those cases
On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice@freeswitch.org <http://krice@freeswitch.org> > wrote: This is probably the same scenario as this is exactly what to expect... Call gets answered far end doesn’t ACK FS sending them a 200OK , fs hangsup the call....
Quite common on networks with NAT issues or broken endpoints
On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100@gmail.com <http://vetali100@gmail.com> <http://vetali100@gmail.com> > wrote:
I saw this happened earlier when the remote party does not send SIP ACK after receiving SIP OK, so the call is being disconnected after exactly 32 seconds. Not sure if this is exact same scenario here, but just something to consider...
Regards. Vitalie
2012/11/15 kaleem rehman <k4kaleem@gmail.com <http://k4kaleem@gmail.com> <http://k4kaleem@gmail.com> > Hi All,
my inbound calls are fine with no issues, my outbound calls get disconnected after 32 seconds and its on all calls. i tried 2 different suppliers and its same result. please find the attached log file with sofia in debug mode. - caller was extension 1234 and desination was 01908321682
your help will be greately appreciated.
regards, Kaleem
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