The problem with stock libss7 is that one will never complete the tests required from telcos in Europe as it is missing functionality which ITU have described in the test procedures. Without the ISUP timers (the main functionality added from the patches) it is just not possible and the link may not even come UP in some cases. Probably in ANSI world it works fine, but not in the ITU world. One of the difficulties was to keep the code working as before without the timers defined in chan_dahdi.conf, another is the hard to find (freely available) ANSI standard and it's requirements, then the code base/functionality have changed quite a lot from 1.6 separating sig_ss7 from chan_dahdi etc. I am sure there are bugs and place for improvements for that branch, but the original version from Domjan is used from me and many more for few years already (that's why i said the bugs are from me in that branch :) ) and we are stuck with 1.6 because of that. I have tried to get the changes in Asterisk 11, but my (below average) C skills and available time did not allowed to do that, while the more time passes more difficult it will be to keep it up to date with the rest of the Asterisk code. I hope with some help from others (testing and patches) this code will finally find it's way in Asterisk and then we may look to adding the cluster/routing/STP functionality

On 2013-06-26 16:33, Marcelo Pacheco wrote:
Thanks Kaloyan.
Before this thread, there were no mentions at all to a KNK tree, so I
though this was stock libss7.
I'm using my own patched libss7.
I processed over one million call setups with ten servers, with many
difficult setups (third party STPs, STPs 1000 miles away with
transmission lines that do fail from time to time, connections with
almost one dozen types of ISUP switches, sharing two links with an STP
to half a dozen switches), and the issues reported don't happen at all.
So this look like a bug in your patch or in Attila's code.
My patch is for paying customers only (they get the source, and could
release if they want to, but chose not to).
I have done very small changes to the ISUP side of things, but some
fairly major changes to MTP2, MTP3 and DAHDI mtp2 mode.
I even implemented very basic STP functionality, and MTP2 over UDP
signalling (between asterisks only).
It might be worth trying to look at the diffs from stock to your branch.

On 06/26/13 09:05, Kaloyan Kovachev wrote:
Almost forgot. Please do not post patches (if any) in this list, but
attach them to the SS7-27 issue instead with proper license agreement,
so it can be included in Asterisk codebase

On 2013-06-26 14:57, Kaloyan Kovachev wrote:
Hi all,
sorry for joining so late, but i am on holidays (by the end of the
week) and rarely checking my mailbox. Thanks to bad weather i did that
today :)

To the OP:
while reading the first posts i thought it is an old problem with
REL/RSC loop (persistent on start with ANSI signaling) which was fixed
in libss7 instead of sig_ss7, but not sure if it is a similar yet
different one or it is the same issue. It really is a (remaining)
problem if we receive RLC on previous REL, but after we have sent RSC.
I was thinking to clear the old status bits after we receive RLC, but
this will not fix the double RLC received problem and we can't ignore
the first one (or just clear the SENT_REL flag), because we may never
get a second one, so it should probably be better to ignore sending
second RSC inside isup_handle_unexpected() if the previous one was
sent T17 (timer seconds) ago. Because the timer is stopped on RLC it
should be another timer or some flag to ignore it's expiration and not
reset again ... will work on this next week when i am back.

The code in my branch is actually Domjan Attila's version (the patches
attached to the SS7-27 issue) ported to later Asterisk versions with
very few additions/modifications, so the muffins are for him, while
the bugs are from me :)

P.S.
apologies for top posting - the connection is unstable and i had to
write the post offline and just copy/paste it

On 2013-06-26 06:42, Pavel Troller wrote:
Hi!
So, I'm replying to my own original post, to keep the question and a
possible answer together without any excessive or unrelated information.
I hope I've found the cause of the problem and I hope I solved it. A
modified libss7 is now online and I'm waiting for busy hours to see,
whether
it will help.
The problem is, that in the isup_rel() function, all the important
got_sent_msg flags are cleared, so the stack "forgets" a preceding call
state:
... isup_rel():
c->got_sent_msg |= ISUP_SENT_REL;
c->got_sent_msg &= ~(ISUP_SENT_IAM | ISUP_PENDING_IAM |
ISUP_CALL_CONNECTED | ISUP_GOT_IAM | ISUP_GOT_CCR | ISUP_SENT_INR);
...
So, an incoming MSU, which was perfectly legitimate before sending REL,
is now handled as unexpected.
My solution adds the following code to the isup_receive() function for
every message, which can confuse the stack by the discovered cause
(an example for ACM message):
case ISUP_ACM:
+                       if (c->got_sent_msg & ISUP_SENT_REL) {
+                               ss7_message(ss7, "Got unexpected ACM
after sending REL on CIC %d PC %d, ignoring ", c->cic, opc);
+                               return 0;
+                       }

if (!(c->got_sent_msg & ISUP_SENT_IAM)) {
ss7_message(ss7, "Got ACM but we didn't send IAM on CIC %d PC %d ",
c->cic, opc);
return isup_handle_unexpected(ss7, c, opc);
}

If my change will prove good, I'm planning to remove the
ss7_message() to
limit the stack verbosity, as these situations are relatively
frequent under
heavy load and I think they are moreless logical and normal.

I would be glad for some words from the KNK branch maintainer(s),
whether to
create a JIRA issue and put my patch there or how to proceed now in
general.

With regards,
Pavel



Hi!
I would like to share my expiernce with deployment of this
experimental SS7
branch.
The first impressions are good, especially the timers seem to work well,
saving many calls from being frozen.
However, there are still some strange things, which I would like to
discuss
here, one by one.
The first one is, that the channel sometimes doesn't recognize a message
(mostly RLC), even it comes from an action initiated by the channel
itself.
Typically, the following is appearing often:

[Jun 24 13:33:41] ERROR[3975]: chan_dahdi.c:14406 dahdi_ss7_error:
[1] ISUP timer t17 expired on CIC 27 DPC 4097
[1] Got RLC but we didn't send REL/RSC on CIC 27 PC 4097 reseting the
cic

As I understand, there were some timeouts and now the channel tries to
recover by sending RSC and firing T17. However, it seems that it
immediately
rejects RLC, which comes back as a response to the RSC which was just
sent
upon expiry of T17. And this appears again and again in the rhythm of
T17,
and the channel is not operational.
ss7 show calls shows the following line for the misbehaving CIC:
27  4097  11  IAM                       IAM

Or, a very similar situation:
[2] Got SUS but no call on CIC 48 PC 4096 reseting the CIC
[2] Got RLC but we didn't send REL/RSC on CIC 48 PC 4096 reseting the
CIC

The first question is, why there was no call while SUS was received. My
idea is, that both the parties hung up their phones in the same time and that the call was undergoing destruction on Asterisk side (REL just sent or something like this), while SUS arrived. Maybe the call was marked as
cleared even before RLC came back ? OK, I can understand this. But
if the CIC was reset as the first message says (i.e. RSC was sent),
why the
RLC going back is not recognized then ?

Or, just now the following appeared:

[1] Got ACM but we didn't send IAM on CIC 10 PC 4097 reseting the cic
[1] Got RLC but we didn't send REL/RSC on CIC 10 PC 4097 reseting the
cic

Again, it's questionable, why this happened, but the second line seems
to indicate some brokeness again.

To explain: The channel is operating on a gateway equipped with 16 E1s
and current traffic is about 10 CAPS, there are two linksets to two
cooperating exchanges. They are EWSDs, which have very mature and stable
SS7, so I'm almost sure that they are not making signalling errors.

With regards,
Pavel

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-ss7

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-ss7

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-ss7

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-ss7



--
Atenciosamente,

Marcelo Pacheco
M2J Comunicações e Informática
Fixo: (27)2222-8118 / (27)2233-2296
Vivo: (27)9964-5440
Claro: (27)9312-5319
MSN: marc...@macp.eti.br
E-mail: marc...@m2j.com.br


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-ss7

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-ss7

Reply via email to