30 Apr
2010
30 Apr
'10
6:32 p.m.
Guys,
I'm successfully using a Kamailio + RTPproxy setup in bridge mode with most
of my Gateways. My setup includes two different interfaces one with a public
IP and teh other with the private IP.
Now I'm facing some slight issue. Some providers won't accept my calls (or
calls will have some strange behavior) if the Contact header has an IP out
of immediate range.
I tried to use fix_nated_contact() function but as per my topology, this
function will not change the contact header because the IP is already the
one on the interface.
Example:
U 192.168.200.X:5060 -> 192.168.200.Y:5060
INVITE sip:111160911097@192.168.200.Y SIP/2.0.
Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport.
From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
U 200.A.A.A:5060 -> 200.B.B.B:5060
INVITE sip:898960911097@200.B.B.B SIP/2.0.
Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>.
Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>.
Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0.
Via: SIP/2.0/UDP
192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060.
From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
Is there any way to let know Kamailio the outgoing IP I'll be using and fix
the contact accordingly?
I can trigger this change after I know the destination IP.
Thanks!
Uriel
30 Apr
30 Apr
8:03 p.m.
Hello,
it might not be the solution, because they should route based on
Record-Route headers, not on Contact header. Anyhow changing the Contact
will break the routing, so you will need to store somehow the original
contact.
You can do manual detection in case you do bridging, by checking the
receiving interface, $Ri is the local IP where the request was received,
therfore you will be sending on the other interface. Are you doing force
send socket to select outgoing interface? If yes, then is where you know
the local ip for sending.
Cheers,
Daniel
On 4/30/10 6:32 PM, Uriel Rozenbaum wrote:
Guys,
I'm successfully using a Kamailio + RTPproxy setup in bridge mode with
most of my Gateways. My setup includes two different interfaces one
with a public IP and teh other with the private IP.
Now I'm facing some slight issue. Some providers won't accept my calls
(or calls will have some strange behavior) if the Contact header has
an IP out of immediate range.
I tried to use fix_nated_contact() function but as per my topology,
this function will not change the contact header because the IP is
already the one on the interface.
Example:
U 192.168.200.X:5060 -> 192.168.200.Y:5060
INVITE sip:111160911097@192.168.200.Y SIP/2.0.
Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport.
From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
U 200.A.A.A:5060 -> 200.B.B.B:5060
INVITE sip:898960911097@200.B.B.B SIP/2.0.
Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>.
Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>.
Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0.
Via: SIP/2.0/UDP
192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060.
From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
Is there any way to let know Kamailio the outgoing IP I'll be using
and fix the contact accordingly?
I can trigger this change after I know the destination IP.
Thanks!
Uriel
_______________________________________________
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/danielconstantinmierla
10:10 p.m.
Daniel,
Maybe my question is silly, but in this case the contact should remain
intact? (I mean in bridge mode).
I understand the destination UA should read the Record-route Headers and
ignore the contents of the Contact Header, but I think this is not what´s
happening.
I'm not using force_socket because the gateway already knows how to route
the calls and I'm detecting the outgoing interface before calling
force_rtp_proxy with flags.
Should I replace the contact using REGEX?
Thanks,
Uriel
On Fri, Apr 30, 2010 at 3:03 PM, Daniel-Constantin Mierla <miconda(a)gmail.com
wrote:
Hello,
it might not be the solution, because they should route based on
Record-Route headers, not on Contact header. Anyhow changing the Contact
will break the routing, so you will need to store somehow the original
contact.
You can do manual detection in case you do bridging, by checking the
receiving interface, $Ri is the local IP where the request was received,
therfore you will be sending on the other interface. Are you doing force
send socket to select outgoing interface? If yes, then is where you know the
local ip for sending.
Cheers,
Daniel
On 4/30/10 6:32 PM, Uriel Rozenbaum wrote:
Guys,
I'm successfully using a Kamailio + RTPproxy setup in bridge mode with most
of my Gateways. My setup includes two different interfaces one with a public
IP and teh other with the private IP.
Now I'm facing some slight issue. Some providers won't accept my calls (or
calls will have some strange behavior) if the Contact header has an IP out
of immediate range.
I tried to use fix_nated_contact() function but as per my topology, this
function will not change the contact header because the IP is already the
one on the interface.
Example:
U 192.168.200.X:5060 -> 192.168.200.Y:5060
INVITE sip:111160911097@192.168.200.Y SIP/2.0.
Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport.
From: "Uriel Rozenbaum"
<sip:60911100@192.168.200.X><sip:60911100@192.168.200.X>
;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y> <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
U 200.A.A.A:5060 -> 200.B.B.B:5060
INVITE sip:898960911097@200.B.B.B SIP/2.0.
Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>.
Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>.
Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0.
Via: SIP/2.0/UDP
192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060.
From: "Uriel Rozenbaum"
<sip:60911100@192.168.200.X><sip:60911100@192.168.200.X>
;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y> <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
Is there any way to let know Kamailio the outgoing IP I'll be using and fix
the contact accordingly?
I can trigger this change after I know the destination IP.
Thanks!
Uriel
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://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/danielconstantinmierla
10:26 p.m.
It sucks to deal with broken SIP implementation on the carrier side.
Here's one thing that you can try. You can encode the contact:
http://kamailio.org/docs/modules/stable/modules_k/siputils.html#id2596873
Like this you will have the public IP in the Contact header. Then,
for all subsequent in-dialog request, you check if the RURI is encoded
and you decode and then route normally.
http://kamailio.org/docs/modules/stable/modules_k/siputils.html#id2587362
I was using this method with success with several switches.
Hope this helps.
Regards,
Ovidiu Sas
On Fri, Apr 30, 2010 at 4:10 PM, Uriel Rozenbaum
<uriel.rozenbaum(a)gmail.com> wrote:
Daniel,
Maybe my question is silly, but in this case the contact should remain
intact? (I mean in bridge mode).
I understand the destination UA should read the Record-route Headers and
ignore the contents of the Contact Header, but I think this is not what´s
happening.
I'm not using force_socket because the gateway already knows how to route
the calls and I'm detecting the outgoing interface before calling
force_rtp_proxy with flags.
Should I replace the contact using REGEX?
Thanks,
Uriel
On Fri, Apr 30, 2010 at 3:03 PM, Daniel-Constantin Mierla
<miconda(a)gmail.com> wrote:
Hello,
it might not be the solution, because they should route based on
Record-Route headers, not on Contact header. Anyhow changing the Contact
will break the routing, so you will need to store somehow the original
contact.
You can do manual detection in case you do bridging, by checking the
receiving interface, $Ri is the local IP where the request was received,
therfore you will be sending on the other interface. Are you doing force
send socket to select outgoing interface? If yes, then is where you know the
local ip for sending.
Cheers,
Daniel
On 4/30/10 6:32 PM, Uriel Rozenbaum wrote:
Guys,
I'm successfully using a Kamailio + RTPproxy setup in bridge mode with
most of my Gateways. My setup includes two different interfaces one with a
public IP and teh other with the private IP.
Now I'm facing some slight issue. Some providers won't accept my calls (or
calls will have some strange behavior) if the Contact header has an IP out
of immediate range.
I tried to use fix_nated_contact() function but as per my topology, this
function will not change the contact header because the IP is already the
one on the interface.
Example:
U 192.168.200.X:5060 -> 192.168.200.Y:5060
INVITE sip:111160911097@192.168.200.Y SIP/2.0.
Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport.
From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@192.168.200.X>.
U 200.A.A.A:5060 -> 200.B.B.B:5060
INVITE sip:898960911097@200.B.B.B SIP/2.0.
Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>.
Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>.
Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0.
Via: SIP/2.0/UDP
192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060.
From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@192.168.200.X>.
Is there any way to let know Kamailio the outgoing IP I'll be using and
fix the contact accordingly?
I can trigger this change after I know the destination IP.
Thanks!
Uriel
_______________________________________________
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/danielconstantinmierla
_______________________________________________
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
3 May
3 May
10:47 a.m.
The best would be to tell the providers that they should fix their systems.
As a workaround you could try encode/decode contact functions:
http://sip-router.org/docbook/sip-router/branch/master/modules_k/siputils/s…
IIRC these functions may cause strange results if you do other contact
rewriting too (e.g. fix_nated_contact).
regards
klaus
Am 30.04.2010 18:32, schrieb Uriel Rozenbaum:
Guys,
I'm successfully using a Kamailio + RTPproxy setup in bridge mode with
most of my Gateways. My setup includes two different interfaces one with
a public IP and teh other with the private IP.
Now I'm facing some slight issue. Some providers won't accept my calls
(or calls will have some strange behavior) if the Contact header has an
IP out of immediate range.
I tried to use fix_nated_contact() function but as per my topology, this
function will not change the contact header because the IP is already
the one on the interface.
Example:
U 192.168.200.X:5060 -> 192.168.200.Y:5060
INVITE sip:111160911097@192.168.200.Y SIP/2.0.
Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport.
From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
U 200.A.A.A:5060 -> 200.B.B.B:5060
INVITE sip:898960911097@200.B.B.B SIP/2.0.
Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>.
Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>.
Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0.
Via: SIP/2.0/UDP
192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060.
From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
Is there any way to let know Kamailio the outgoing IP I'll be using and
fix the contact accordingly?
I can trigger this change after I know the destination IP.
Thanks!
Uriel
_______________________________________________
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 May
4 May
3:41 p.m.
Thanks Guys, I'll be trying this. Do you know if I can use AVP or vars as
parameters for these functions?
encode_contact(encoding_prefix)
decode_contact()
On Mon, May 3, 2010 at 5:47 AM, Klaus Darilion <klaus.mailinglists(a)pernau.at
wrote:
The best would be to tell the providers that they
should fix their systems.
As a workaround you could try encode/decode contact functions:
http://sip-router.org/docbook/sip-router/branch/master/modules_k/siputils/s…
IIRC these functions may cause strange results if you do other contact
rewriting too (e.g. fix_nated_contact).
regards
klaus
Am 30.04.2010 18:32, schrieb Uriel Rozenbaum:
Guys,
I'm successfully using a Kamailio + RTPproxy setup in bridge mode with
most of my Gateways. My setup includes two different interfaces one with
a public IP and teh other with the private IP.
Now I'm facing some slight issue. Some providers won't accept my calls
(or calls will have some strange behavior) if the Contact header has an
IP out of immediate range.
I tried to use fix_nated_contact() function but as per my topology, this
function will not change the contact header because the IP is already
the one on the interface.
Example:
U 192.168.200.X:5060 -> 192.168.200.Y:5060
INVITE sip:111160911097@192.168.200.Y SIP/2.0.
Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport.
From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
U 200.A.A.A:5060 -> 200.B.B.B:5060
INVITE sip:898960911097@200.B.B.B SIP/2.0.
Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>.
Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>.
Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0.
Via: SIP/2.0/UDP
192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060.
From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
Is there any way to let know Kamailio the outgoing IP I'll be using and
fix the contact accordingly?
I can trigger this change after I know the destination IP.
Thanks!
Uriel
_______________________________________________
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
6:25 p.m.
I have no idea ... :-(
Am 04.05.2010 15:41, schrieb Uriel Rozenbaum:
Thanks Guys, I'll be trying this. Do you know if I
can use AVP or vars
as parameters for these functions?
|encode_contact(encoding_prefix)
||decode_contact()|
On Mon, May 3, 2010 at 5:47 AM, Klaus Darilion
<klaus.mailinglists(a)pernau.at <mailto:klaus.mailinglists@pernau.at>> wrote:
The best would be to tell the providers that they should fix their
systems.
As a workaround you could try encode/decode contact functions:
http://sip-router.org/docbook/sip-router/branch/master/modules_k/siputils/s…
IIRC these functions may cause strange results if you do other
contact rewriting too (e.g. fix_nated_contact).
regards
klaus
Am 30.04.2010 18:32, schrieb Uriel Rozenbaum:
Guys,
I'm successfully using a Kamailio + RTPproxy setup in bridge
mode with
most of my Gateways. My setup includes two different interfaces
one with
a public IP and teh other with the private IP.
Now I'm facing some slight issue. Some providers won't accept my
calls
(or calls will have some strange behavior) if the Contact header
has an
IP out of immediate range.
I tried to use fix_nated_contact() function but as per my
topology, this
function will not change the contact header because the IP is
already
the one on the interface.
Example:
U 192.168.200.X:5060 -> 192.168.200.Y:5060
INVITE sip:111160911097@192.168.200.Y SIP/2.0.
Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport.
From: "Uriel Rozenbaum"
<sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
U 200.A.A.A:5060 -> 200.B.B.B:5060
INVITE sip:898960911097@200.B.B.B SIP/2.0.
Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>.
Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>.
Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0.
Via: SIP/2.0/UDP
192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060.
From: "Uriel Rozenbaum"
<sip:60911100@192.168.200.X>;tag=as32794d5e.
To: <sip:111160911097@192.168.200.Y>.
Contact: <sip:60911100@*192.168.200.X*>.
Is there any way to let know Kamailio the outgoing IP I'll be
using and
fix the contact accordingly?
I can trigger this change after I know the destination IP.
Thanks!
Uriel
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list
sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
7:18 p.m.
No AVPs or vars. The encoding field separator can be specified via params:
http://sip-router.org/docbook/sip-router/branch/master/modules_k/siputils/s…
Regards,
Ovidiu Sas
On Tue, May 4, 2010 at 9:41 AM, Uriel Rozenbaum
<uriel.rozenbaum(a)gmail.com> wrote:
Thanks Guys, I'll be trying this. Do you know if I
can use AVP or vars as
parameters for these functions?
encode_contact(encoding_prefix)
decode_contact()
On Mon, May 3, 2010 at 5:47 AM, Klaus Darilion
<klaus.mailinglists(a)pernau.at> wrote:
The best would be to tell the providers that they should fix their
systems.
As a workaround you could try encode/decode contact functions:
http://sip-router.org/docbook/sip-router/branch/master/modules_k/siputils/s…
IIRC these functions may cause strange results if you do other contact
rewriting too (e.g. fix_nated_contact).
regards
klaus
Am 30.04.2010 18:32, schrieb Uriel Rozenbaum:
>
> Guys,
>
> I'm successfully using a Kamailio + RTPproxy setup in bridge mode with
> most of my Gateways. My setup includes two different interfaces one with
> a public IP and teh other with the private IP.
>
> Now I'm facing some slight issue. Some providers won't accept my calls
> (or calls will have some strange behavior) if the Contact header has an
> IP out of immediate range.
>
> I tried to use fix_nated_contact() function but as per my topology, this
> function will not change the contact header because the IP is already
> the one on the interface.
>
> Example:
> U 192.168.200.X:5060 -> 192.168.200.Y:5060
> INVITE sip:111160911097@192.168.200.Y SIP/2.0.
> Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport.
> From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
> To: <sip:111160911097@192.168.200.Y>.
> Contact: <sip:60911100@*192.168.200.X*>.
>
> U 200.A.A.A:5060 -> 200.B.B.B:5060
> INVITE sip:898960911097@200.B.B.B SIP/2.0.
> Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>.
> Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>.
> Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0.
> Via: SIP/2.0/UDP
>
> 192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060.
> From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e.
> To: <sip:111160911097@192.168.200.Y>.
> Contact: <sip:60911100@*192.168.200.X*>.
>
> Is there any way to let know Kamailio the outgoing IP I'll be using and
> fix the contact accordingly?
> I can trigger this change after I know the destination IP.
>
> Thanks!
> Uriel
>
>
>
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7 comments
4 participants
participants (4)
-
Daniel-Constantin Mierla -
Klaus Darilion -
Ovidiu Sas -
Uriel Rozenbaum