Hi,
What would be the best way to generate and send an INVITE message generated by the Kamailio server. This would be for a new dialog session, UAS initiated (e.g. to make a click 2 call like service).
I've tried the $uac_req(method)="INVITE"; uac_req_send(); but didn't work. I gather something would be needed first to open the dialogue before doing it but couldn't find it on the dialogue module.
Thanks,
Joao Alves
This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp
Hello Joao,
As a general rule, speaking broadly and methodologically, Kamailio - as a SIP proxy - is not a good vehicle for the initiation of calls, since it cannot, itself, be a party to a dialog. Synthesising an INVITE using uac_req_send() isn't going to work because there's no second party to the dialog. Where will the replies to the INVITE go? Kamailio itself cannot be a party.
There are, nevertheless, the ways to initiate a call from Kamailio using the 'dialog' module. The most straightforward is probably the 'dlg_bridge' / 'dlg.bridge' (MI and RPC respectively) commands, which can be called externally to Kamailio,
http://kamailio.org/docs/modules/4.3.x/modules/dialog.html#idp3756384
or internally from route script, via the dlg_bridge() command:
http://kamailio.org/docs/modules/4.3.x/modules/dialog.html#idp3698368
The general idea here is that an INVITE is initiated to one party and then that party is REFER'd out to another party. REFER is a SIP method that is commonly used for unattended transfers and call forwarding. You can read more about it here:
https://www.ietf.org/rfc/rfc3515.txt
Of course, for it to work, party #1 has to support REFER.
-- Alex
Hi Alex,
I do realize Kamailio to be a SIP proxy and, as you said, Kamailio may not be the best place to do a HTTP <> SIP gateway. I start to realize as time goes by...:) However, at least managing incoming SIP sessions and converting those to HTTP using http_query() although didn't go very smooth, but at the end, results are quite positive.
Now was starting doing the other direction HTTP=> SIP and using xhttp_reply() noticed was not so much available to start a dialogue and hence this email - as new starter in kamailio there are many modules and functions that not aware of and this asking to the expert's community.
Tell me one thing. If was not a SIP INVITE but just a SIP INFO or SIP MESSAGE (since are stateless), would it be easier to do?
I'll be off next week, but I'll comeback to this the week after.
All inputs are welcome!
Thanks Joao
-----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Alex Balashov Sent: sexta-feira, 19 de Junho de 2015 20:28 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] New SIP INVITE from UAS (new dialogue)
Hello Joao,
As a general rule, speaking broadly and methodologically, Kamailio - as a SIP proxy - is not a good vehicle for the initiation of calls, since it cannot, itself, be a party to a dialog. Synthesising an INVITE using uac_req_send() isn't going to work because there's no second party to the dialog. Where will the replies to the INVITE go? Kamailio itself cannot be a party.
There are, nevertheless, the ways to initiate a call from Kamailio using the 'dialog' module. The most straightforward is probably the 'dlg_bridge' / 'dlg.bridge' (MI and RPC respectively) commands, which can be called externally to Kamailio,
http://kamailio.org/docs/modules/4.3.x/modules/dialog.html#idp3756384
or internally from route script, via the dlg_bridge() command:
http://kamailio.org/docs/modules/4.3.x/modules/dialog.html#idp3698368
The general idea here is that an INVITE is initiated to one party and then that party is REFER'd out to another party. REFER is a SIP method that is commonly used for unattended transfers and call forwarding. You can read more about it here:
https://www.ietf.org/rfc/rfc3515.txt
Of course, for it to work, party #1 has to support REFER.
-- Alex
-- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.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
This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp
On 06/19/2015 04:30 PM, Joao Alves wrote:
Tell me one thing. If was not a SIP INVITE but just a SIP INFO or SIP MESSAGE (since are stateless), would it be easier to do?
Yes, those would be more viable.
Hi again,
Following up on this and having done some further tests, I was finally able to generate a SIP INVITE request using the following command sequence on the command file.
$uac_req(method)="INVITE"; $uac_req(ruri)=$shv(userAcontact); $uac_req(turi)="sip:1000@decasa.website"; $uac_req(furi)="sip:1001@decasa.website"; $uac_req(callid)=$var(call_id);
$uac_req(hdrs)="Contact: <sip:1001@151.236.52.84:5060;transport=udp\r\n"; $uac_req(hdrs)="Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE\r\n"; $uac_req(hdrs)="Content-Type: application/sdp\r\n"; $uac_req(body)=$var(offer);
uac_req_send();
I've attached the trace collected on the client side. Transaction/dialogue wise seems to have the correct behavior, however I found two blocking issues which are causing the SIP INVITE to be rejected with a 400 "bad request" error code.
The first, it seems we cannot send more than one header on the SIP INVITE. That is I need to add the "Contact", "Allow" and "Content-Type" fields, but only the last one is inserted, which is the "Content-Type". If I marked it as a comment (with #), them the "Allow" is included instead.
The second issue is related with the inclusion of SDP offer on the body of the message. Since this SDP is quite a long one(1266 bytes) it is being truncated.
I'm currently using kamailio 4.3.0.
Thanks, Joao
-----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Alex Balashov Sent: sexta-feira, 19 de Junho de 2015 22:11 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] New SIP INVITE from UAS (new dialogue)
On 06/19/2015 04:30 PM, Joao Alves wrote:
Tell me one thing. If was not a SIP INVITE but just a SIP INFO or SIP MESSAGE (since are stateless), would it be easier to do?
Yes, those would be more viable.
-- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.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
This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp
On Wednesday 22 July 2015 12:37:58 Joao Alves wrote:
The first, it seems we cannot send more than one header on the SIP INVITE. That is I need to add the "Contact", "Allow" and "Content-Type" fields, but only the last one is inserted, which is the "Content-Type". If I marked it as a comment (with #), them the "Allow" is included instead.
A quick solution could be: $uac_req(hdrs)="Contact:..\r\nAllow:..\r\nContent-type: application/sdp\r\n";
The second issue is related with the inclusion of SDP offer on the body of the message. Since this SDP is quite a long one(1266 bytes) it is being truncated.
Where? How did you check this?
Hi Daniel,
For the first issue, your solution did work!
In relation with the SDP size, I originally just compared with the source one (see attached). What I just did was to double check using an online tool and confirmed that the original has 1266 bytes (as also indicated by the Content's length) while the one sent has only 1077 bytes.
Many Thanks, Joao
-----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Daniel Tryba Sent: quarta-feira, 22 de Julho de 2015 13:58 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] New SIP INVITE from UAS (new dialogue)
On Wednesday 22 July 2015 12:37:58 Joao Alves wrote:
The first, it seems we cannot send more than one header on the SIP INVITE. That is I need to add the "Contact", "Allow" and "Content-Type" fields, but only the last one is inserted, which is the "Content-Type". If I marked it as a comment (with #), them the "Allow" is included instead.
A quick solution could be: $uac_req(hdrs)="Contact:..\r\nAllow:..\r\nContent-type: application/sdp\r\n";
The second issue is related with the inclusion of SDP offer on the body of the message. Since this SDP is quite a long one(1266 bytes) it is being truncated.
Where? How did you check this?
_______________________________________________ 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
This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp
On Wednesday 22 July 2015 14:01:02 Joao Alves wrote:
In relation with the SDP size, I originally just compared with the source one (see attached). What I just did was to double check using an online tool and confirmed that the original has 1266 bytes (as also indicated by the Content's length) while the one sent has only 1077 bytes.
I saw the capture after my message. The message is fragmented and not shown correctly in the capture. There shouldn't be any problem (unless there are network issues between you and destination).
Hi Daniel,
Yes, you're right. It was fragmented at UDP level. I've repeated with TCP as transport and the SDP is complete.
Thanks again, Joao
-----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Daniel Tryba Sent: quarta-feira, 22 de Julho de 2015 16:39 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] New SIP INVITE from UAS (new dialogue)
On Wednesday 22 July 2015 14:01:02 Joao Alves wrote:
In relation with the SDP size, I originally just compared with the source one (see attached). What I just did was to double check using an online tool and confirmed that the original has 1266 bytes (as also indicated by the Content's length) while the one sent has only 1077 bytes.
I saw the capture after my message. The message is fragmented and not shown correctly in the capture. There shouldn't be any problem (unless there are network issues between you and destination).
_______________________________________________ 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
This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp
Hi,
Doing further step into this, I have realized that despite the SIP INVITE being sent, I cannot create/associate a new dialogue or transaction that I'll need for managing the SIP session.
So following the uac_req_send() I tried to do:
t_newtran(); record_route();
dlg_manage(); if(is_known_dlg()) { xlog("Request $rm from $ci is in-dialog\n"); }
9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=972 a=24 n=uac_req_send 9(44916) DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=sip:1000@188.165.231.30:12060;rtcweb-breaker=no;transport=udp;ws-src-ip=81.84.0.236;ws-src-port=1025;ws-src-proto=ws 9(44916) DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 39161 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=973 a=24 n=t_newtran 9(44916) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff 9(44916) ERROR: tm [t_lookup.c:453]: t_lookup_request(): ERROR: TM module: t_lookup_request: too few headers 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=974 a=24 n=record_route 9(44916) ERROR: <core> [parser/parse_from.c:53]: parse_from_header(): ERROR:parse_from_header: bad msg or missing FROM header 9(44916) ERROR: rr [record.c:407]: record_route(): From parsing failed 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=976 a=24 n=dlg_manage 9(44916) ERROR: dialog [dlg_handlers.c:1561]: dlg_manage(): bad TO header 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=985 a=16 n=if 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=977 a=24 n=is_known_dlg 9(44916) ERROR: dialog [dlg_handlers.c:652]: pre_match_parse(): bad request or missing CALLID/TO hdr :-/
Am I doing something wrong here, as it seems the headers fields are just being "hardcoded" on the SIP INVITE are not updated on the context for that new outbound session (and thus causing the typical dialogue/transaction functions to fail...).
I realized already from one early comment I got from Alex Balashov that this is not a typical usecase for kamailio (being a sip proxy), but is this a show stopper?
Thanks, Joao -----Original Message----- From: Joao Alves Sent: quarta-feira, 22 de Julho de 2015 17:02 To: sr-users@lists.sip-router.org Subject: RE: [SR-Users] New SIP INVITE from UAS (new dialogue)
Hi Daniel,
Yes, you're right. It was fragmented at UDP level. I've repeated with TCP as transport and the SDP is complete.
Thanks again, Joao
-----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Daniel Tryba Sent: quarta-feira, 22 de Julho de 2015 16:39 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] New SIP INVITE from UAS (new dialogue)
On Wednesday 22 July 2015 14:01:02 Joao Alves wrote:
In relation with the SDP size, I originally just compared with the source one (see attached). What I just did was to double check using an online tool and confirmed that the original has 1266 bytes (as also indicated by the Content's length) while the one sent has only 1077 bytes.
I saw the capture after my message. The message is fragmented and not shown correctly in the capture. There shouldn't be any problem (unless there are network issues between you and destination).
_______________________________________________ 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
This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp
Hello,
uac_req_send() is not creating a request that is available in config file afterwards -- that function is sending out the request immediately and that is, the config is processing whatever was there before uac_req_send().
There are functions to send requests parts of dialog, but not exposed to config file, there are inside tm module. For example, presence or dialog modules are using those functions to send NOTIFY or BYE requests, which need to within dialogs created by SUBSCRIBE or INVITE.
In other words, you need to write some C code to expose those functions to config file or write a new module in C to implement the functionality you need, replying internally on tm module (like it is done by presence or dialog).
Cheers, Daniel
On 24/07/15 19:28, Joao Alves wrote:
Hi,
Doing further step into this, I have realized that despite the SIP INVITE being sent, I cannot create/associate a new dialogue or transaction that I'll need for managing the SIP session.
So following the uac_req_send() I tried to do:
t_newtran(); record_route();
dlg_manage(); if(is_known_dlg()) { xlog("Request $rm from $ci is in-dialog\n"); }
9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=972 a=24 n=uac_req_send 9(44916) DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=sip:1000@188.165.231.30:12060;rtcweb-breaker=no;transport=udp;ws-src-ip=81.84.0.236;ws-src-port=1025;ws-src-proto=ws 9(44916) DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 39161 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=973 a=24 n=t_newtran 9(44916) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff 9(44916) ERROR: tm [t_lookup.c:453]: t_lookup_request(): ERROR: TM module: t_lookup_request: too few headers 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=974 a=24 n=record_route 9(44916) ERROR: <core> [parser/parse_from.c:53]: parse_from_header(): ERROR:parse_from_header: bad msg or missing FROM header 9(44916) ERROR: rr [record.c:407]: record_route(): From parsing failed 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=976 a=24 n=dlg_manage 9(44916) ERROR: dialog [dlg_handlers.c:1561]: dlg_manage(): bad TO header 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=985 a=16 n=if 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=977 a=24 n=is_known_dlg 9(44916) ERROR: dialog [dlg_handlers.c:652]: pre_match_parse(): bad request or missing CALLID/TO hdr :-/
Am I doing something wrong here, as it seems the headers fields are just being "hardcoded" on the SIP INVITE are not updated on the context for that new outbound session (and thus causing the typical dialogue/transaction functions to fail...).
I realized already from one early comment I got from Alex Balashov that this is not a typical usecase for kamailio (being a sip proxy), but is this a show stopper?
Thanks, Joao -----Original Message----- From: Joao Alves Sent: quarta-feira, 22 de Julho de 2015 17:02 To: sr-users@lists.sip-router.org Subject: RE: [SR-Users] New SIP INVITE from UAS (new dialogue)
Hi Daniel,
Yes, you're right. It was fragmented at UDP level. I've repeated with TCP as transport and the SDP is complete.
Thanks again, Joao
-----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Daniel Tryba Sent: quarta-feira, 22 de Julho de 2015 16:39 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] New SIP INVITE from UAS (new dialogue)
On Wednesday 22 July 2015 14:01:02 Joao Alves wrote:
In relation with the SDP size, I originally just compared with the source one (see attached). What I just did was to double check using an online tool and confirmed that the original has 1266 bytes (as also indicated by the Content's length) while the one sent has only 1077 bytes.
I saw the capture after my message. The message is fragmented and not shown correctly in the capture. There shouldn't be any problem (unless there are network issues between you and destination).
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
This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp
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
-
Alex Balashov -
Daniel Tryba -
Daniel-Constantin Mierla -
Joao Alves