5 Dec
2013
5 Dec
'13
3:47 p.m.
Hi,
I locally generate BYE using dlg_end_dlg. When I want to end a call that is
"transport layer" bridged, the BYE is not sent to first hop in route_set
but directly to the endpoint. In such BYE there are no Route headers. In
non-bridging calls Routes are correctly placed and the message is routed to
the first "hop".
When the error happens, this is written to a log:
WARNING: rr [loose.c:821]: after_loose(): no socket found for match second
RR
Here ([SR-Users] no socket found for match second RR) I have read this is
only a warning, but in my configuration it seriously influences the message
routing.
My setup is
phone1(192.168.10.3) <--TCP--> kamailio1(192.168.10.2) <--UDP-->
kamailio2(192.168.5.3) <--UDP--> phone2
On kamailio1 I generate dlg_end_dlg and the BYE is sent to phone1 and
phone2 directly.
I'm using Kamailio 4.0.4 on Debian machines.
How can I make the Kamailio1 to send the BYE to kamailio2 in the transport
layer bridging scenario? Do I have some misconfiguration or this is not a
correct behaviour?
Thanks for answer
Efelin
11 Dec
11 Dec
9:50 a.m.
Hello,
can you share the database record or kamctl mi dlg_list for such call?
It helps to see if record route values are stored properly in first place.
Cheers,
Daniel
On 05/12/13 14:47, Efelin Novak wrote:
Hi,
I locally generate BYE using dlg_end_dlg. When I want to end a call
that is "transport layer" bridged, the BYE is not sent to first hop in
route_set but directly to the endpoint. In such BYE there are no Route
headers. In non-bridging calls Routes are correctly placed and the
message is routed to the first "hop".
When the error happens, this is written to a log:
WARNING: rr [loose.c:821]: after_loose(): no socket found for match
second RR
Here ([SR-Users] no socket found for match second RR) I have read this
is only a warning, but in my configuration it seriously influences the
message routing.
My setup is
phone1(192.168.10.3) <--TCP--> kamailio1(192.168.10.2) <--UDP-->
kamailio2(192.168.5.3) <--UDP--> phone2
On kamailio1 I generate dlg_end_dlg and the BYE is sent to phone1 and
phone2 directly.
I'm using Kamailio 4.0.4 on Debian machines.
How can I make the Kamailio1 to send the BYE to kamailio2 in the
transport layer bridging scenario? Do I have some misconfiguration or
this is not a correct behaviour?
Thanks for answer
Efelin
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)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
10:30 a.m.
I just tested this with dialog timeout and can confirm that there are no
Route headers in the locally generated BYEs.
The Record-Route set for the dialog is:
Record-Route:
<sip:172.30.105.18;lr=on;ftag=ervXH3ycHcgpK;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA--;proxy_media=yes;dlgcor=fa8.9ea>
The BYEs from Kamailio don't have a Route header at all. Just these:
03:27:37.464071 IP 172.30.105.18.5060 > 172.30.105.20.5060: SIP, length: 367
E.......@._'.4..?..U.....w^.BYE sip:mod_sofia@172.30.105.20:5060 SIP/2.0
Via: SIP/2.0/UDP
172.30.105.18;branch=z9hG4bK6e86.3c0a9a04000000000000000000000000.0
To: <sip:6785551212@172.30.105.20>;tag=ervXH3ycHcgpK
From:
<sip:4045551212@172.30.105.18>;tag=SD7hk6c99-ac3f4687+1+337b0015+5485357b
CSeq: 1 BYE
Call-ID: 34fdb244-dce1-1231-ee8e-00163e6490b7
Content-Length: 0
Max-Forwards: 70
Of course, that doesn't stop them from working.
-- Alex
On 12/11/2013 02:50 AM, Daniel-Constantin Mierla wrote:
Hello,
can you share the database record or kamctl mi dlg_list for such call?
It helps to see if record route values are stored properly in first place.
Cheers,
Daniel
On 05/12/13 14:47, Efelin Novak wrote:
--
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,
I locally generate BYE using dlg_end_dlg. When I want to end a call
that is "transport layer" bridged, the BYE is not sent to first hop in
route_set but directly to the endpoint. In such BYE there are no Route
headers. In non-bridging calls Routes are correctly placed and the
message is routed to the first "hop".
When the error happens, this is written to a log:
WARNING: rr [loose.c:821]: after_loose(): no socket found for match
second RR
Here ([SR-Users] no socket found for match second RR) I have read this
is only a warning, but in my configuration it seriously influences the
message routing.
My setup is
phone1(192.168.10.3) <--TCP--> kamailio1(192.168.10.2) <--UDP-->
kamailio2(192.168.5.3) <--UDP--> phone2
On kamailio1 I generate dlg_end_dlg and the BYE is sent to phone1 and
phone2 directly.
I'm using Kamailio 4.0.4 on Debian machines.
How can I make the Kamailio1 to send the BYE to kamailio2 in the
transport layer bridging scenario? Do I have some misconfiguration or
this is not a correct behaviour?
Thanks for answer
Efelin
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
10:36 a.m.
Hi,
why should Kamailio add a route for himself to an outbound request?
172.30.105.18 is obviously the proxy itself; so adding the
Route-Header makes no sense.... (at least from a SIP-Perspective).
Record-Route:
<sip:172.30.105.18;lr=on;ftag=ervXH3ycHcgpK;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA--;proxy_media=yes;dlgcor=fa8.9ea>
03:27:37.464071 IP 172.30.105.18.5060 > 172.30.105.20.5060: SIP, length: 367
Or am i missing something here?
Kind regards,
Carsten
2013/12/11 Alex Balashov <abalashov(a)evaristesys.com>om>:
I just tested this with dialog timeout and can confirm
that there are no
Route headers in the locally generated BYEs.
The Record-Route set for the dialog is:
Record-Route:
<sip:172.30.105.18;lr=on;ftag=ervXH3ycHcgpK;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA--;proxy_media=yes;dlgcor=fa8.9ea>
The BYEs from Kamailio don't have a Route header at all. Just these:
03:27:37.464071 IP 172.30.105.18.5060 > 172.30.105.20.5060: SIP, length: 367
E.......@._'.4..?..U.....w^.BYE sip:mod_sofia@172.30.105.20:5060 SIP/2.0
Via: SIP/2.0/UDP
172.30.105.18;branch=z9hG4bK6e86.3c0a9a04000000000000000000000000.0
To: <sip:6785551212@172.30.105.20>;tag=ervXH3ycHcgpK
From:
<sip:4045551212@172.30.105.18>;tag=SD7hk6c99-ac3f4687+1+337b0015+5485357b
CSeq: 1 BYE
Call-ID: 34fdb244-dce1-1231-ee8e-00163e6490b7
Content-Length: 0
Max-Forwards: 70
Of course, that doesn't stop them from working.
-- Alex
On 12/11/2013 02:50 AM, Daniel-Constantin Mierla wrote:
--
Carsten Bock
CEO (Geschäftsführer)
ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany
http://www.ng-voice.com
mailto:carsten@ng-voice.com
Office +49 40 34927219
Fax +49 40 34927220
Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284
Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/
Hello,
can you share the database record or kamctl mi dlg_list for such call?
It helps to see if record route values are stored properly in first place.
Cheers,
Daniel
On 05/12/13 14:47, Efelin Novak wrote:
--
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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi,
I locally generate BYE using dlg_end_dlg. When I want to end a call
that is "transport layer" bridged, the BYE is not sent to first hop in
route_set but directly to the endpoint. In such BYE there are no Route
headers. In non-bridging calls Routes are correctly placed and the
message is routed to the first "hop".
When the error happens, this is written to a log:
WARNING: rr [loose.c:821]: after_loose(): no socket found for match
second RR
Here ([SR-Users] no socket found for match second RR) I have read this
is only a warning, but in my configuration it seriously influences the
message routing.
My setup is
phone1(192.168.10.3) <--TCP--> kamailio1(192.168.10.2) <--UDP-->
kamailio2(192.168.5.3) <--UDP--> phone2
On kamailio1 I generate dlg_end_dlg and the BYE is sent to phone1 and
phone2 directly.
I'm using Kamailio 4.0.4 on Debian machines.
How can I make the Kamailio1 to send the BYE to kamailio2 in the
transport layer bridging scenario? Do I have some misconfiguration or
this is not a correct behaviour?
Thanks for answer
Efelin
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
10:38 a.m.
Carsten,
On 12/11/2013 03:36 AM, Carsten Bock wrote:
why should Kamailio add a route for himself to an
outbound request?
172.30.105.18 is obviously the proxy itself; so adding the
Route-Header makes no sense.... (at least from a SIP-Perspective).
Record-Route:
<sip:172.30.105.18;lr=on;ftag=ervXH3ycHcgpK;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA--;proxy_media=yes;dlgcor=fa8.9ea>
03:27:37.464071 IP 172.30.105.18.5060 > 172.30.105.20.5060: SIP, length: 367
Or am i missing something here?
That depends, philosophically, on whether the intention of this feature
is to spoof BYEs so that they appear to come from the respective UAs
toward each other through the proxy (or, in substance, act as if they
were), or for Kamailio to unexpectedly take on the role of a UAC
mid-call. :-)
-- Alex
--
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/
10:41 a.m.
Hello,
On 11/12/13 09:38, Alex Balashov wrote:
Carsten,
On 12/11/2013 03:36 AM, Carsten Bock wrote:
there must not be any route for proxy itself, even when BYE
is sent by
end UA the Route is consumed by proxy, so it is no difference from this
perspective.
In the initial email on this thread, there was a second proxy, so I
wanted to see the route header stored for that proxy.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
why should Kamailio add a route for himself to an
outbound request?
172.30.105.18 is obviously the proxy itself; so adding the
Route-Header makes no sense.... (at least from a SIP-Perspective).
Record-Route:
<sip:172.30.105.18;lr=on;ftag=ervXH3ycHcgpK;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA--;proxy_media=yes;dlgcor=fa8.9ea>
03:27:37.464071 IP 172.30.105.18.5060 > 172.30.105.20.5060: SIP,
length: 367
Or am i missing something here?
That depends, philosophically, on whether the intention of this
feature is to spoof BYEs so that they appear to come from the
respective UAs toward each other through the proxy (or, in substance,
act as if they were), or for Kamailio to unexpectedly take on the role
of a UAC mid-call. :-)
10:49 a.m.
On 12/11/2013 03:41 AM, Daniel-Constantin Mierla wrote:
there must not be any route for proxy itself, even
when BYE is sent by
end UA the Route is consumed by proxy, so it is no difference from this
perspective.
Oh, yeah. [slaps head] That's true. You and Carsten are correct.
-- Alex
--
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/
3 Jan
3 Jan
6:52 p.m.
Hi Daniel,
thanks for an answer & sorry for the delay (Christmas time).
Here we go:
Scenario: phone1(192.168.9.3, 192.168.10.75 (nat)) <--TCP-->
kamailio1(192.168.10.2) <--UDP--> kamailio2(192.168.5.3) <--UDP-->
phone2(192.168.12.2)
dialog:: hash=1456:1090244220
state:: 4
timestart:: 1388767148
timeout:: 111170022
callid:: 7fbca98b-22ea049(a)192.168.9.3
from_uri:: sip:phone1@example.com
from_tag:: 17cfe8a323b2ac11o1
caller_contact:: sip:phone1@192.168.10.75:5075;transport=tcp
caller_cseq:: 102
caller_route_set::
caller_bind_addr:: tcp:192.168.10.2:5060
to_uri:: sip:phone2@example.com
to_tag:: a94c095b773be1dd6e8d668a785a9c84ec059e7d
callee_contact:: sip:phone2@192.168.12.2:5060
callee_cseq:: 102
callee_route_set::
<sip:192.168.10.2;r2=on;lr=on;ftag=17cfe8a323b2ac11o1;vsf=AAAAAAAGAQIIAAIDAgh5UEBXIR0VSRIfZS5zaw--;nat=yes;did=0b5.c7ecbf04>,<sip:192.168.5.3;lr=on;did=0b5.0b>
callee_bind_addr:: udp:192.168.10.2:5060
Efelin
2013/12/11 Alex Balashov <abalashov(a)evaristesys.com>
On 12/11/2013 03:41 AM, Daniel-Constantin Mierla
wrote:
there must not be any route for proxy itself, even when BYE is sent by
end UA the Route is consumed by proxy, so it is
no difference from this
perspective.
Oh, yeah. [slaps head] That's true. You and Carsten are correct.
-- Alex
--
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/
_______________________________________________
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7 comments
4 participants
participants (4)
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Alex Balashov -
Carsten Bock -
Daniel-Constantin Mierla -
Efelin Novak