3 Mar
2017
3 Mar
'17
3:53 a.m.
Greetings,
I have a general question about the usage of SIP contact headers in the
context of using Kamailio as a SIP proxy.
________________________________
[ A ] --> [ Kamailio B ] ---> [ C ]
Node A originates a SIP invite, containing a valid via header and URI while
setting the contact address to a user@itself and delivers it to Kamailio B
which is acting as a SIP proxy.
Kamailio B then uses dispatcher routing to direct the Invite to node C,
adding a via line with its own information as well as a record-route header
with its own proxy information but retaining the same contact address from
A.
Node C establishes the call and then sends a re-invite to the Kamailio B
proxy which is in turn sent to Node A. Node A responds with a 200 OK
The problem arises when Node C tries to send an Ack in response to this 200
OK. The ack is being sent to the Contact address, rather than the routing
already established in the initial dialog.
________________________________
My question is: should kamailio be rewriting this contact address with its
own? Is that the best practice? My understanding is that the contact header
is more so related to future requests within the same dialog ONLY when a
record-route is not used.
I would appreciate any clarification on the RFC or best practices in this
scenario.
Thank you,
Jack Davis
3 Mar
3 Mar
4:03 a.m.
I seem to recall reading something stating that you shouldn't modify the
contact header - But, I can't find the text.
Might suggest adding a Record-Route header instead of re-writing the
contact header?
I'm curious as to what the list has to say about this question, I've
considered doing the same thing.
On 03/02/2017 04:53 PM, Jack Davis wrote:
Greetings,
I have a general question about the usage of SIP contact headers in the
context of using Kamailio as a SIP proxy.
________________________________
[ A ] --> [ Kamailio B ] ---> [ C ]
Node A originates a SIP invite, containing a valid via header and URI
while setting the contact address to a user@itself and delivers it to
Kamailio B which is acting as a SIP proxy.
Kamailio B then uses dispatcher routing to direct the Invite to node C,
adding a via line with its own information as well as a record-route
header with its own proxy information but retaining the same contact
address from A.
Node C establishes the call and then sends a re-invite to the Kamailio B
proxy which is in turn sent to Node A. Node A responds with a 200 OK
The problem arises when Node C tries to send an Ack in response to this
200 OK. The ack is being sent to the Contact address, rather than the
routing already established in the initial dialog.
________________________________
My question is: should kamailio be rewriting this contact address with
its own? Is that the best practice? My understanding is that the contact
header is more so related to future requests within the same dialog ONLY
when a record-route is not used.
I would appreciate any clarification on the RFC or best practices in
this scenario.
Thank you,
Jack Davis
_______________________________________________
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
4:14 a.m.
No, Kamailio should definitely not be rewriting the Contact. That is set
by the remote dialog parties only.
If the ACK is being sent directly to the host in the domain portion of
the Contact address, that is because the endpoint is ignoring/not
honouring Record-Route. That's radioactively wrong behaviour and should
be fixed.
Reinvites do not alter the Route set, as clearly stated in RFC 3261 §
12.2:
Requests within a dialog MAY contain Record-Route and Contact header
fields. However, these requests do not cause the dialog's route set
to be modified, although they may modify the remote target URI.
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
11:14 p.m.
For a SIP proxy, the preferred method is indeed to add a Record-Route
In the script, add record_route(); instruction.
Emmanuel
IVèS
Le 2017-03-02 à 20:03, Robert Johnson a écrit :
I seem to recall reading something stating that you
shouldn't modify
the contact header - But, I can't find the text.
Might suggest adding a Record-Route header instead of re-writing the
contact header?
I'm curious as to what the list has to say about this question, I've
considered doing the same thing.
On 03/02/2017 04:53 PM, Jack Davis wrote:
Greetings,
I have a general question about the usage of SIP contact headers in the
context of using Kamailio as a SIP proxy.
________________________________
[ A ] --> [ Kamailio B ] ---> [ C ]
Node A originates a SIP invite, containing a valid via header and URI
while setting the contact address to a user@itself and delivers it to
Kamailio B which is acting as a SIP proxy.
Kamailio B then uses dispatcher routing to direct the Invite to node C,
adding a via line with its own information as well as a record-route
header with its own proxy information but retaining the same contact
address from A.
Node C establishes the call and then sends a re-invite to the Kamailio B
proxy which is in turn sent to Node A. Node A responds with a 200 OK
The problem arises when Node C tries to send an Ack in response to this
200 OK. The ack is being sent to the Contact address, rather than the
routing already established in the initial dialog.
________________________________
My question is: should kamailio be rewriting this contact address with
its own? Is that the best practice? My understanding is that the contact
header is more so related to future requests within the same dialog ONLY
when a record-route is not used.
I would appreciate any clarification on the RFC or best practices in
this scenario.
Thank you,
Jack Davis
_______________________________________________
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
_______________________________________________
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
11:35 p.m.
Thank you all for the replies.
I'm already utilizing a record-route header in the initial Invite before
any re-invites occur.
Is the preferred method to also include a record-route header in the 200 OK
that is given in response to the re-invite?
Thank you,
Jack Davis
On Fri, Mar 3, 2017 at 2:14 PM, Emmanuel BUU <emmanuel.buu(a)ives.fr> wrote:
For a SIP proxy, the preferred method is indeed to add
a Record-Route
In the script, add record_route(); instruction.
Emmanuel
IVèS
Le 2017-03-02 à 20:03, Robert Johnson a écrit :
I seem to recall reading something stating that
you shouldn't modify the
contact header - But, I can't find the text.
Might suggest adding a Record-Route header instead of re-writing the
contact header?
I'm curious as to what the list has to say about this question, I've
considered doing the same thing.
On 03/02/2017 04:53 PM, Jack Davis wrote:
_______________________________________________
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
Greetings,
I have a general question about the usage of SIP contact headers in the
context of using Kamailio as a SIP proxy.
________________________________
[ A ] --> [ Kamailio B ] ---> [ C ]
Node A originates a SIP invite, containing a valid via header and URI
while setting the contact address to a user@itself and delivers it to
Kamailio B which is acting as a SIP proxy.
Kamailio B then uses dispatcher routing to direct the Invite to node C,
adding a via line with its own information as well as a record-route
header with its own proxy information but retaining the same contact
address from A.
Node C establishes the call and then sends a re-invite to the Kamailio B
proxy which is in turn sent to Node A. Node A responds with a 200 OK
The problem arises when Node C tries to send an Ack in response to this
200 OK. The ack is being sent to the Contact address, rather than the
routing already established in the initial dialog.
________________________________
My question is: should kamailio be rewriting this contact address with
its own? Is that the best practice? My understanding is that the contact
header is more so related to future requests within the same dialog ONLY
when a record-route is not used.
I would appreciate any clarification on the RFC or best practices in
this scenario.
Thank you,
Jack Davis
_______________________________________________
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
_______________________________________________
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
11:39 p.m.
On Fri, Mar 03, 2017 at 02:35:10PM -0600, Jack Davis wrote:
Is the preferred method to also include a record-route
header in the 200 OK
that is given in response to the re-invite?
No. Only initial INVITEs and their replies should have a Record-Route
header. In-dialog requests and their replies contain Route headers that
are constructed from that Record-Route set.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
8 Mar
8 Mar
12:44 a.m.
What kind of problems can arise from inserting said header?
I'm trying to find a way to get a pesky setup to start behaving, of course,
without requiring them to make any changes.
Thank you,
Jack Davis
On Fri, Mar 3, 2017 at 2:39 PM, Alex Balashov <abalashov(a)evaristesys.com>
wrote:
On Fri, Mar 03, 2017 at 02:35:10PM -0600, Jack Davis
wrote:
Is the preferred method to also include a
record-route header in the 200
OK
that is given in response to the re-invite?
No. Only initial INVITEs and their replies should have a Record-Route
header. In-dialog requests and their replies contain Route headers that
are constructed from that Record-Route set.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.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
12:49 a.m.
On Tue, Mar 07, 2017 at 03:44:31PM -0600, Jack Davis wrote:
What kind of problems can arise from inserting said
header?
Theoretically, none. It should be ignored, as per the RFC. Moreover, the
RFC doesn't _prohibit_ such a header in an in-dialog request, so you are
probably clear to add it. Fundamentally, § 12.2 says:
Requests within a dialog MAY contain Record-Route and Contact header
fields. However, these requests do not cause the dialog's route set
to be modified...
I have certainly seen instances in the field where RRs in in-dialog
requests are not ignored, however. That's on the endpoints, of course,
but it can make things worse.
I'm trying to find a way to get a pesky setup to
start behaving, of course,
without requiring them to make any changes.
That's always a tough gig.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
15 Mar
15 Mar
1:44 a.m.
Thank you for the replies, this has been helpful.
So if I do want to go down this road and want to practically implement a
solution, what is the best method of appending additional record route
headers to a 200 OK response?
I guess, specifically, adding them to the 200 OK response that is given by
kamailio to a re-invite.
Thank you,
Jack Davis
On Tue, Mar 7, 2017 at 3:49 PM, Alex Balashov <abalashov(a)evaristesys.com>
wrote:
On Tue, Mar 07, 2017 at 03:44:31PM -0600, Jack Davis
wrote:
What kind of problems can arise from inserting
said header?
Theoretically, none. It should be ignored, as per the RFC. Moreover, the
RFC doesn't _prohibit_ such a header in an in-dialog request, so you are
probably clear to add it. Fundamentally, § 12.2 says:
Requests within a dialog MAY contain Record-Route and Contact header
fields. However, these requests do not cause the dialog's route set
to be modified...
I have certainly seen instances in the field where RRs in in-dialog
requests are not ignored, however. That's on the endpoints, of course,
but it can make things worse.
I'm trying to find a way to get a pesky setup
to start behaving, of
course,
without requiring them to make any changes.
That's always a tough gig.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.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
1:46 a.m.
append_to_reply("Record-Route: <sip:...>\r\n");
But I'd like to advise against it once more. :)
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
2:06 a.m.
Thanks Alex.
And objection duly noted, =) I'll just have to test it out and make sure it
doesn't break any of my other clients.
Thank you,
Jack Davis
On Tue, Mar 14, 2017 at 5:46 PM, Alex Balashov <abalashov(a)evaristesys.com>
wrote:
append_to_reply("Record-Route:
<sip:...>\r\n");
But I'd like to advise against it once more. :)
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
1:02 p.m.
Hello,
I haven't followed closely the entire discussion, but for proper
clarification, if one needs to add a header to a received SIP response
that is forwarded, then it has to use append_hf(...) inside on/reply_route.
The append_to_reply() should be used for SIP responses to be generated
by Kamailio.
Cheers,
Daniel
On 14/03/2017 23:46, Alex Balashov wrote:
append_to_reply("Record-Route:
<sip:...>\r\n");
But I'd like to advise against it once more. :)
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________
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--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
6:02 p.m.
On Wed, Mar 15, 2017 at 11:02:22AM +0100, Daniel-Constantin Mierla wrote:
I haven't followed closely the entire discussion,
but for proper
clarification, if one needs to add a header to a received SIP response
that is forwarded, then it has to use append_hf(...) inside on/reply_route.
The append_to_reply() should be used for SIP responses to be generated
by Kamailio.
This is absolutely true, and entirely the result of poor memory on my
part. For some reason I misremembered that the OP was trying to generate
a synthetic reply internally with attributes of the received reply.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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days inactive
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12 comments
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participants (5)
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Alex Balashov -
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Emmanuel BUU -
Jack Davis -
Robert Johnson