Hoho,
i want to implement RFC3578 (transaction oriented) in Kamailio. Therefore I have to correlate two different Invites with same Call-ID. I see various ways of accomplishing this, but I am stuck at all of my solutions. Find the algorithm I want to implement attached.
Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached)
The following lines, I'm going to explain my thoughts. For reference I'm calling "Thread A" the first INVITE coming in and "Thread B" the second.
####################### ### using Module ASYNC * Thread A -> async_route("handle_overlap","<timer t99 value>"); * Thread B -> if new incoming transaction, ** set avp("<call-id>")= <digitlength of called number> and ** call async_route("handle_overlap","<timer t99 value>"); * in route[handle_overlap] ** if avp("call-id") greater than "<digitlength of called number>" *** cancel transaction with "484 -Address Incomplete" ** if avp("call-id") equals "<digitlength of called number>" *** handle call => the number is complete PROBLEM: The old transaction is not canceled immediately, but after t99 elapsed...
####################### ### using Module TMX using t_suspend() and t_continue(), but here i see the following problems: * Thread A calls "t_suspend()" ** How is it possible to cancel "Thread A"'s transaction from Thread B? ** If there is no Thread B, so the numbers were complete, how can I get Thread A to continue processing?? * General TMX-Question: Are there any examples available?
####################### ### using simple sleep() and shv()/avp() Sounds like dirty hacking... I'd appriciate a solution with one of aboves modules. But if there is a solution available, I'd try this.
Thx in advise Moritz
On 04 Feb 2014, at 15:43, Moritz Graf moritz.graf@g-fit.de wrote:
Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached)
You are not allowed to have two open INVITEs, the second one SHOULD get a 491 response. The client should not send a new INVITE if the old INVITE transaction is not complete.
I don't think overlap dialing in SIP was ever meant to be timer based. Consider that the first INVITE can go to a different proxy than the second. There's no dialog, no route set.
/O
Indeed, Section 3.1 in the referenced RFC covers this pretty well.
On 4 February 2014 09:47:38 GMT-05:00, "Olle E. Johansson" oej@edvina.net wrote:
On 04 Feb 2014, at 15:43, Moritz Graf moritz.graf@g-fit.de wrote:
Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If
not,
you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response
and
start the timer again. (Find the algorithm I want to implement
attached) You are not allowed to have two open INVITEs, the second one SHOULD get a 491 response. The client should not send a new INVITE if the old INVITE transaction is not complete.
I don't think overlap dialing in SIP was ever meant to be timer based. Consider that the first INVITE can go to a different proxy than the second. There's no dialog, no route set.
/O _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Sent from my Nexus 10, with all the figments of autocorrect that might imply.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Hi,
sure. That's why it's called "transaction oriented". On a SIP-Theory basis, the second INVITE opens a new transaction!! For correlation the Call-ID keeps the same! This is how all of the big carriers I know handle OverlapDialing.
The problem is: * I am receiving an INVITE. * At this point I don't know if there will be a second INVITE or not. So I have to wait. * If the timer elapses, ok handle this INVITE. * But if there IS a second (third..) INVITE, I have to cancel the first and then handle the second.
Sounds like nobody has done this before... any suggetions appriciated.
greetz
Am 04.02.2014 15:56, schrieb Alex Balashov:
Indeed, Section 3.1 in the referenced RFC covers this pretty well.
On 4 February 2014 09:47:38 GMT-05:00, "Olle E. Johansson" oej@edvina.net wrote:
On 04 Feb 2014, at 15:43, Moritz Graf <moritz.graf@g-fit.de> wrote: Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached) You are not allowed to have two open INVITEs, the second one SHOULD get a 491 response. The client should not send a new INVITE if the old INVITE transaction is not complete. I don't th ink overlap dialing in SIP was ever meant to be timer based. Consider that the first INVITE can go to a different proxy than the second. There's no dialog, no route set. /O ------------------------------------------------------------------------ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-- Sent from my Nexus 10, with all the figments of autocorrect that might imply.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
I'm not sure this is a task for a proxy like Kamailio, even with its fancy new async relay features. To truly customise how successive INVITEs are dealt with like this, I think it needs to be handled by a UAS that has complete latitude in terms of how it computes state.
On 4 February 2014 10:13:33 GMT-05:00, Moritz Graf moritz.graf@g-fit.de wrote:
Hi,
sure. That's why it's called "transaction oriented". On a SIP-Theory basis, the second INVITE opens a new transaction!! For correlation the Call-ID keeps the same! This is how all of the big carriers I know handle OverlapDialing.
The problem is:
- I am receiving an INVITE.
- At this point I don't know if there will be a second INVITE or not.
So I have to wait.
- If the timer elapses, ok handle this INVITE.
- But if there IS a second (third..) INVITE, I have to cancel the first
and then handle the second.
Sounds like nobody has done this before... any suggetions appriciated.
greetz
Am 04.02.2014 15:56, schrieb Alex Balashov:
Indeed, Section 3.1 in the referenced RFC covers this pretty well.
On 4 February 2014 09:47:38 GMT-05:00, "Olle E. Johansson" oej@edvina.net wrote:
On 04 Feb 2014, at 15:43, Moritz Graf <moritz.graf@g-fit.de>wrote:
Shortly explained what RFC3578 is: In a open numbering planyou
never know if the INVITE you received is already complete, or ifthere are
more numbers coming in. One way of accomplishing this is toset up a
timer. If the timer elapses you assume the number iscomplete.
If not, you are receiving a new INVITE with one digit more. Now youhave to
close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want toimplement
attached) You are not allowed to have two open INVITEs, the second oneSHOULD get
a 491 response. The client should not send a new INVITE if theold INVITE
transaction is not complete. I don't th ink overlap dialing in SIP was ever meant to be timer based. Consider that the first INVITE can go to a different proxy than thesecond. There's no
dialog, no route set. /O
SIP Express Router (SER) and Kamailio (OpenSER) - sr-usersmailing list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-- Sent from my Nexus 10, with all the figments of autocorrect that
might
imply.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
--
Moritz Graf, B.Sc. Betrieb NGN-Plattform
G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg
Telefon +49 (9 41) 69 85 - 1 86 Telefax +49 (9 41) 69 85 - 2 86 mailto:moritz.graf@g-fit.de http://www.g-fit.de
G-FIT Gesellschaft für innovative Telekommunikationsdienste mbH & Co. KG, Kommanditgesellschaft, Sitz Regensburg, Registergericht Regensburg, HRA 7626; Geschäftsführer: Dipl.Inf. (FH) Alfred Rauscher
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Sent from my Nexus 10, with all the figments of autocorrect that might imply.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Hi,
nah, that's not what I wanted to hear ;). And it's not really complicated. I hacked it down (my third choice in the initial mail) with $shv(). See (find also attached):
#################################################### ### Hacky section xdbgl("Shared Variable of current ( Call-ID = $ci ) is $shv($(ci)) "); $var(numberLength) = $(var(calledNumber_normalized){s.len}); xdbgl("Length of string ( $var(calledNumber_normalized) )is $var(numberLength) characters."); if ( $shv($(ci)) < $var(numberLength) ) { xdbgl("I detected, that the old value is lower, setting it to the new value."); $shv($(ci)) = $var(numberLength); } # sleeping to wait for potentially new invites sl_send_reply("100", "Trying..."); # So no retransmits happen usleep("500000"); #half a second, this is the timer i was speaking about if( $var(numberLength) < $shv($(ci)) ) { xdbgl("Detected newer INVITE, aborting this transaction with 484 response"); route("address_incomplete"); break; } ######################################################
Do you see any problems with this?
So this is not really smooth, probably someone has a better solution for this??
greetz
Am 04.02.2014 16:16, schrieb Alex Balashov:
I'm not sure this is a task for a proxy like Kamailio, even with its fancy new async relay features. To truly customise how successive INVITEs are dealt with like this, I think it needs to be handled by a UAS that has complete latitude in terms of how it computes state.
On 4 February 2014 10:13:33 GMT-05:00, Moritz Graf moritz.graf@g-fit.de wrote:
Hi, sure. That's why it's called "transaction oriented". On a SIP-Theory basis, the second INVITE opens a new transaction!! For correlation the Call-ID keeps the same! This is how all of the big carriers I know handle OverlapDialing. The problem is: * I am receiving an INVITE. * At this point I don't know if there will be a second INVITE or not. So I have to wait. * If the timer elapses, ok handle this INVITE. * But if there IS a second (third..) INVITE, I have to cancel the first and then handle the second. Sounds like nobody has done this before... any suggetions appriciated. greetz Am 04.02.2014 15:56, schrieb Alex Balashov: Indeed, Section 3.1 in the referenced RFC covers this pretty well. On 4 February 2014 09:47 :38 GMT-05:00, "Olle E. Johansson" <oej@edvina.net> wrote: On 04 Feb 2014, at 15:43, Moritz Graf <moritz.graf@g-fit.de> wrote: Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached) You are not allowed to have two open INVITEs, the second one SHOULD get a 491 response. The client should not send a new INVITE if the ol d INVITE transaction is not complete. I don't th ink overlap dialing in SIP was ever meant to be timer based. Consider that the first INVITE can go to a different proxy than the second. There's no dialog, no route set. /O ------------------------------------------------------------------------ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sent from my Nexus 10, with all the figments of autocorrect that might imply. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/-- Sent from my Nexus 10, with all the figments of autocorrect that might imply.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Again, you are trying to break the SIP Protocol. Check RFC 3261 for references. There should only be one SIP invite open at a time. That's the reason the SIP community invented UPDATE.
And you assume a lot of things - that your invites are sent to the same server, which may not be the case. DNS lookups and load balancing/failover is done on a per transaction basis.
The timing is usually done with local dialplans and stuff in the user agents. The proxy can answer each invite individually and implement overlap support, but the timing should be left to the user agent.
/O
On 04 Feb 2014, at 17:47, Moritz Graf moritz.graf@g-fit.de wrote:
Hi,
nah, that's not what I wanted to hear ;). And it's not really complicated. I hacked it down (my third choice in the initial mail) with $shv(). See (find also attached):
#################################################### ### Hacky section xdbgl("Shared Variable of current ( Call-ID = $ci ) is $shv($(ci)) "); $var(numberLength) = $(var(calledNumber_normalized){s.len}); xdbgl("Length of string ( $var(calledNumber_normalized) )is $var(numberLength) characters."); if ( $shv($(ci)) < $var(numberLength) ) { xdbgl("I detected, that the old value is lower, setting it to the new value."); $shv($(ci)) = $var(numberLength); } # sleeping to wait for potentially new invites sl_send_reply("100", "Trying..."); # So no retransmits happen usleep("500000"); #half a second, this is the timer i was speaking about if( $var(numberLength) < $shv($(ci)) ) { xdbgl("Detected newer INVITE, aborting this transaction with 484 response"); route("address_incomplete"); break; } ######################################################
Do you see any problems with this?
So this is not really smooth, probably someone has a better solution for this??
greetz
Am 04.02.2014 16:16, schrieb Alex Balashov:
I'm not sure this is a task for a proxy like Kamailio, even with its fancy new async relay features. To truly customise how successive INVITEs are dealt with like this, I think it needs to be handled by a UAS that has complete latitude in terms of how it computes state.
On 4 February 2014 10:13:33 GMT-05:00, Moritz Graf moritz.graf@g-fit.de wrote:
Hi,
sure. That's why it's called "transaction oriented". On a SIP-Theory basis, the second INVITE opens a new transaction!! For correlation the Call-ID keeps the same! This is how all of the big carriers I know handle OverlapDialing.
The problem is:
- I am receiving an INVITE.
- At this point I don't know if there will be a second INVITE or not. So
I have to wait.
- If the timer elapses, ok handle this INVITE.
- But if there IS a second (third..) INVITE, I have to cancel the first
and then handle the second.
Sounds like nobody has done this before... any suggetions appriciated.
greetz
Am 04.02.2014 15:56, schrieb Alex Balashov:
Indeed, Section 3.1 in the referenced RFC covers this pretty well. On 4 February 2014 09:47 :38 GMT-05:00, "Olle E. Johansson" <oej@edvina.net> wrote: On 04 Feb 2014, at 15:43, Moritz Graf <moritz.graf@g-fit.de> wrote: Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached) You are not allowed to have two open INVITEs, the second one SHOULD get a 491 response. The client should not send a new INVITE if the ol d INVITE transaction is not complete. I don't th ink overlap dialing in SIP was ever meant to be timer based. Consider that the first INVITE can go to a different proxy than the second. There's no dialog, no route set. /O ------------------------------------------------------------------------ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sent from my Nexus 10, with all the figments of autocorrect that might imply. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/-- Sent from my Nexus 10, with all the figments of autocorrect that might imply.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
--
Moritz Graf, B.Sc. Betrieb NGN-Plattform
G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg
Telefon +49 (9 41) 69 85 - 1 86 Telefax +49 (9 41) 69 85 - 2 86 mailto:moritz.graf@g-fit.de http://www.g-fit.de
G-FIT Gesellschaft für innovative Telekommunikationsdienste mbH & Co. KG, Kommanditgesellschaft, Sitz Regensburg, Registergericht Regensburg, HRA 7626; Geschäftsführer: Dipl.Inf. (FH) Alfred Rauscher <handleOverlap.txt>_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 02/04/2014 12:54 PM, Olle E. Johansson wrote:
And you assume a lot of things - that your invites are sent to the same server, which may not be the case. DNS lookups and load balancing/failover is done on a per transaction basis.
In fairness, RFC 3578 does offer a fair bit of commentary on this caveat:
However, having subsequent INVITEs routed in different ways brings some problems as well. The first INVITE, for instance, might be routed to a particular gateway, and a subsequent INVITE, to another. The result is that both gateways generate an IAM. Since one of the IAMs (or both) has an incomplete number, it would fail, having already consumed PSTN resources.
[...]
Routing in SIP can be controlled by the administrator of the network. Therefore, a gateway can be configured to generate SIP overlap signalling in the way described below only if the SIP routing infrastructure ensures that INVITEs will only reach one gateway. When the routing infrastructure is not under the control of the administrator of the gateway, the procedures of Section 2 have to be used instead.
And, while I agree that this is ridiculous and is in conflict with the basic spirit of SIP, somehow 3578 did become an RFC... I am as puzzled by that as you may be. :-)
-- Alex
On 04 Feb 2014, at 18:58, Alex Balashov abalashov@evaristesys.com wrote:
On 02/04/2014 12:54 PM, Olle E. Johansson wrote:
And you assume a lot of things - that your invites are sent to the same server, which may not be the case. DNS lookups and load balancing/failover is done on a per transaction basis.
In fairness, RFC 3578 does offer a fair bit of commentary on this caveat:
However, having subsequent INVITEs routed in different ways brings some problems as well. The first INVITE, for instance, might be routed to a particular gateway, and a subsequent INVITE, to another. The result is that both gateways generate an IAM. Since one of the IAMs (or both) has an incomplete number, it would fail, having already consumed PSTN resources.
[...]
Routing in SIP can be controlled by the administrator of the network. Therefore, a gateway can be configured to generate SIP overlap signalling in the way described below only if the SIP routing infrastructure ensures that INVITEs will only reach one gateway. When the routing infrastructure is not under the control of the administrator of the gateway, the procedures of Section 2 have to be used instead.
And, while I agree that this is ridiculous and is in conflict with the basic spirit of SIP, somehow 3578 did become an RFC... I am as puzzled by that as you may be. :-)
That's applying a strict route. IMS has some similar overrides too... Ouch.
Regardless, it doesn't talk about overlapping INVITE transactions, just a series of INVITE transactions.
/O
Hello,
I haven't read full thread, but I noticed in the config snippet below that you use $shv(...) - this is single value variable that stores the value in shared memory. The $ci is not replaced in definition of $shv($ci), but is used as literal token '$ci'.
You should use htable module, defining a hash table, say 'overlap' and then use $sht(overlap=>$ci).
Cheers, Daniel
On 04/02/14 17:47, Moritz Graf wrote:
Hi,
nah, that's not what I wanted to hear ;). And it's not really complicated. I hacked it down (my third choice in the initial mail) with $shv(). See (find also attached):
#################################################### ### Hacky section xdbgl("Shared Variable of current ( Call-ID = $ci ) is $shv($(ci)) "); $var(numberLength) = $(var(calledNumber_normalized){s.len}); xdbgl("Length of string ( $var(calledNumber_normalized) )is $var(numberLength) characters."); if ( $shv($(ci)) < $var(numberLength) ) { xdbgl("I detected, that the old value is lower, setting it to the new value."); $shv($(ci)) = $var(numberLength); } # sleeping to wait for potentially new invites sl_send_reply("100", "Trying..."); # So no retransmits happen usleep("500000"); #half a second, this is the timer i was speaking about if( $var(numberLength) < $shv($(ci)) ) { xdbgl("Detected newer INVITE, aborting this transaction with 484 response"); route("address_incomplete"); break; } ######################################################
Do you see any problems with this?
So this is not really smooth, probably someone has a better solution for this??
greetz
Am 04.02.2014 16:16, schrieb Alex Balashov:
I'm not sure this is a task for a proxy like Kamailio, even with its fancy new async relay features. To truly customise how successive INVITEs are dealt with like this, I think it needs to be handled by a UAS that has complete latitude in terms of how it computes state.
On 4 February 2014 10:13:33 GMT-05:00, Moritz Graf moritz.graf@g-fit.de wrote:
Hi, sure. That's why it's called "transaction oriented". On a SIP-Theory basis, the second INVITE opens a new transaction!! For correlation the Call-ID keeps the same! This is how all of the big carriers I know handle OverlapDialing. The problem is: * I am receiving an INVITE. * At this point I don't know if there will be a second INVITE or not. So I have to wait. * If the timer elapses, ok handle this INVITE. * But if there IS a second (third..) INVITE, I have to cancel the first and then handle the second. Sounds like nobody has done this before... any suggetions appriciated. greetz Am 04.02.2014 15:56, schrieb Alex Balashov: Indeed, Section 3.1 in the referenced RFC covers this pretty well. On 4 February 2014 09:47 :38 GMT-05:00, "Olle E. Johansson" <oej@edvina.net> wrote: On 04 Feb 2014, at 15:43, Moritz Graf <moritz.graf@g-fit.de> wrote: Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached) You are not allowed to have two open INVITEs, the second one SHOULD get a 491 response. The client should not send a new INVITE if the ol d INVITE transaction is not complete. I don't th ink overlap dialing in SIP was ever meant to be timer based. Consider that the first INVITE can go to a different proxy than the second. There's no dialog, no route set. /O ------------------------------------------------------------------------ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sent from my Nexus 10, with all the figments of autocorrect that might imply. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/-- Sent from my Nexus 10, with all the figments of autocorrect that might imply.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Uff, that could have led to some ugly failures!!
Ur right! That's sth my sipp-tests don't cover...
Thx aaa lot!
greetz
Am 05.02.2014 09:17, schrieb Daniel-Constantin Mierla:
Hello,
I haven't read full thread, but I noticed in the config snippet below that you use $shv(...) - this is single value variable that stores the value in shared memory. The $ci is not replaced in definition of $shv($ci), but is used as literal token '$ci'.
You should use htable module, defining a hash table, say 'overlap' and then use $sht(overlap=>$ci).
Cheers, Daniel
On 04/02/14 17:47, Moritz Graf wrote:
Hi,
nah, that's not what I wanted to hear ;). And it's not really complicated. I hacked it down (my third choice in the initial mail) with $shv(). See (find also attached):
#################################################### ### Hacky section xdbgl("Shared Variable of current ( Call-ID = $ci ) is $shv($(ci)) "); $var(numberLength) = $(var(calledNumber_normalized){s.len}); xdbgl("Length of string ( $var(calledNumber_normalized) )is $var(numberLength) characters."); if ( $shv($(ci)) < $var(numberLength) ) { xdbgl("I detected, that the old value is lower, setting it to the new value."); $shv($(ci)) = $var(numberLength); } # sleeping to wait for potentially new invites sl_send_reply("100", "Trying..."); # So no retransmits happen usleep("500000"); #half a second, this is the timer i was speaking about if( $var(numberLength) < $shv($(ci)) ) { xdbgl("Detected newer INVITE, aborting this transaction with 484 response"); route("address_incomplete"); break; } ######################################################
Do you see any problems with this?
So this is not really smooth, probably someone has a better solution for this??
greetz
Am 04.02.2014 16:16, schrieb Alex Balashov:
I'm not sure this is a task for a proxy like Kamailio, even with its fancy new async relay features. To truly customise how successive INVITEs are dealt with like this, I think it needs to be handled by a UAS that has complete latitude in terms of how it computes state.
On 4 February 2014 10:13:33 GMT-05:00, Moritz Graf moritz.graf@g-fit.de wrote:
Hi, sure. That's why it's called "transaction oriented". On a SIP-Theory basis, the second INVITE opens a new transaction!! For correlation the Call-ID keeps the same! This is how all of the big carriers I know handle OverlapDialing. The problem is: * I am receiving an INVITE. * At this point I don't know if there will be a second INVITE or not. So I have to wait. * If the timer elapses, ok handle this INVITE. * But if there IS a second (third..) INVITE, I have to cancel the first and then handle the second. Sounds like nobody has done this before... any suggetions appriciated. greetz Am 04.02.2014 15:56, schrieb Alex Balashov: Indeed, Section 3.1 in the referenced RFC covers this pretty well. On 4 February 2014 09:47 :38 GMT-05:00, "Olle E. Johansson" <oej@edvina.net> wrote: On 04 Feb 2014, at 15:43, Moritz Graf <moritz.graf@g-fit.de> wrote: Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached) You are not allowed to have two open INVITEs, the second one SHOULD get a 491 response. The client should not send a new INVITE if the ol d INVITE transaction is not complete. I don't th ink overlap dialing in SIP was ever meant to be timer based. Consider that the first INVITE can go to a different proxy than the second. There's no dialog, no route set. /O ------------------------------------------------------------------------ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sent from my Nexus 10, with all the figments of autocorrect that might imply. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/-- Sent from my Nexus 10, with all the figments of autocorrect that might imply.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hi,
so with this heavy use of the usleep(); command i have to set the "children=" parameter to a very high value. Calculating with expected CAPS of 30 and up to 7 subsequent INVITES per call and the sleep of 2 seconds, it's: 20*7*2 = 280
So it has to be children=280
Do you see a Problem with this?
Is there any way to go arround this? -> Is it possible with the ASYNC-Module? But I am not opening a transaction. By now everything is without any transaction, I'm handling this with the sht(callid->$ci) now. -> Is there any example of how to user ASYNC? I can't find one.. :( -> Also is there a example of the usage of the suspend() and continue() functions of tmx?
greetz Moritz
Am 05.02.2014 09:17, schrieb Daniel-Constantin Mierla:
Hello,
I haven't read full thread, but I noticed in the config snippet below that you use $shv(...) - this is single value variable that stores the value in shared memory. The $ci is not replaced in definition of $shv($ci), but is used as literal token '$ci'.
You should use htable module, defining a hash table, say 'overlap' and then use $sht(overlap=>$ci).
Cheers, Daniel
On 04/02/14 17:47, Moritz Graf wrote:
Hi,
nah, that's not what I wanted to hear ;). And it's not really complicated. I hacked it down (my third choice in the initial mail) with $shv(). See (find also attached):
#################################################### ### Hacky section xdbgl("Shared Variable of current ( Call-ID = $ci ) is $shv($(ci)) "); $var(numberLength) = $(var(calledNumber_normalized){s.len}); xdbgl("Length of string ( $var(calledNumber_normalized) )is $var(numberLength) characters."); if ( $shv($(ci)) < $var(numberLength) ) { xdbgl("I detected, that the old value is lower, setting it to the new value."); $shv($(ci)) = $var(numberLength); } # sleeping to wait for potentially new invites sl_send_reply("100", "Trying..."); # So no retransmits happen usleep("500000"); #half a second, this is the timer i was speaking about if( $var(numberLength) < $shv($(ci)) ) { xdbgl("Detected newer INVITE, aborting this transaction with 484 response"); route("address_incomplete"); break; } ######################################################
Do you see any problems with this?
So this is not really smooth, probably someone has a better solution for this??
greetz
Am 04.02.2014 16:16, schrieb Alex Balashov:
I'm not sure this is a task for a proxy like Kamailio, even with its fancy new async relay features. To truly customise how successive INVITEs are dealt with like this, I think it needs to be handled by a UAS that has complete latitude in terms of how it computes state.
On 4 February 2014 10:13:33 GMT-05:00, Moritz Graf moritz.graf@g-fit.de wrote:
Hi, sure. That's why it's called "transaction oriented". On a SIP-Theory basis, the second INVITE opens a new transaction!! For correlation the Call-ID keeps the same! This is how all of the big carriers I know handle OverlapDialing. The problem is: * I am receiving an INVITE. * At this point I don't know if there will be a second INVITE or not. So I have to wait. * If the timer elapses, ok handle this INVITE. * But if there IS a second (third..) INVITE, I have to cancel the first and then handle the second. Sounds like nobody has done this before... any suggetions appriciated. greetz Am 04.02.2014 15:56, schrieb Alex Balashov: Indeed, Section 3.1 in the referenced RFC covers this pretty well. On 4 February 2014 09:47 :38 GMT-05:00, "Olle E. Johansson" <oej@edvina.net> wrote: On 04 Feb 2014, at 15:43, Moritz Graf <moritz.graf@g-fit.de> wrote: Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached) You are not allowed to have two open INVITEs, the second one SHOULD get a 491 response. The client should not send a new INVITE if the ol d INVITE transaction is not complete. I don't th ink overlap dialing in SIP was ever meant to be timer based. Consider that the first INVITE can go to a different proxy than the second. There's no dialog, no route set. /O ------------------------------------------------------------------------ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sent from my Nexus 10, with all the figments of autocorrect that might imply. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/-- Sent from my Nexus 10, with all the figments of autocorrect that might imply.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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Alex Balashov -
Daniel-Constantin Mierla -
Moritz Graf -
Olle E. Johansson