[SR-Users] Question: if SIP Timer L and Timer M from RFC6026 has been defined at Kamailio

[jennifer yang via sr-users]

Hello

Can anyone help answer these questions:

-   if SIP Timer L and Timer M from RFC6026 has been defined at Kamailio 
(version v5.5.x)?
Timer L: Wait time for accepted INVITE  request retransmits
Timer M: Wait time for retransmission of  2xx to INVITE or  additional 2xx from 
 other branches of a forked INVITE
-   if no specific Timer defined, which timer will be used instead?


thanks
br
jenni
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[SR-Users] Question for RFC junkies / kamailio modifying message bodies

[Valentin Christoph via sr-users]

Hi all,

it might be a minor issue, or it might not be an issue at all, but I stumbled 
over following statements in RFC 3261:

Chapter 16.6 Request Forwarding (of a proxy)


  1. Copy request



 The proxy starts with a copy of the received request.  The copy

 MUST initially contain all of the header fields from the

 received request.  Fields not detailed in the processing

 described below MUST NOT be removed.  The copy SHOULD maintain

 the ordering of the header fields as in the received request.

 The proxy MUST NOT reorder field values with a common field

 name (See Section 
7.3.1).  The proxy 
MUST NOT add to, modify,

 or remove the message body.



 An actual implementation need not perform a copy; the primary

 requirement is that the processing for each next hop begin with

 the same request.

Chapter 16.7 Response Processing (at a proxy)


  9.  Forward response



 After performing the processing described in steps "Aggregate

 Authorization Header Field Values" through "Record-Route", the

 proxy MAY perform any feature specific manipulations on the

 selected response.  The proxy MUST NOT add to, modify, or

 remove the message body.  Unless otherwise specified, the proxy

 MUST NOT remove any header field values other than the Via

 header field value discussed in Section 
16.7 Item 3.  In

 particular, the proxy MUST NOT remove any "received" parameter

On the other hand, everywhere in the Internet (starting with stackoverflow), 
you can read it is OK, if a SIP proxy modifies a body, and also kamailio allows 
it.

Might be a question for the coffee break.

All the best
Christoph
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[SR-Users] Question syntax check / plausi check kamailio.cfg

[Valentin Christoph via sr-users]

Dear all,

Is there any other possibility to (automatically) check the plausibility/syntax 
of an existing kamailio.cfg, besides

kamailio -c -f kamailio.cfg

?

Thanks,
Christoph
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[SR-Users] Question about a Slow timer process.

[Володимир Іванець]

Hello everyone!

I'm running Kamailio 5.6.2 and having a potential problem on one of the
systems. The Slow timer process slowly consumes more and more of its PKG
memory. I did not allow it to use all memory by restartink Kamailio during
afterhours so I'm not sure what the consequences might be.

I'm suspecting that this could be caused by my configuration but not sure
how exactly. Can anyone please tell me how the Slow timer process can be
debugged? Is it possible to see what it is storing in the memory and what
data is piling up?

Thanks a lot!

Regarts, Volodymyr Ivanets.
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[SR-Users] Question - Kamailio script and how the return code from a command export is handled as a boolean?

[Chaigneau, Nicolas]

Hello,



I have a question regarding Kamailio script and how the return code from a 
command export is handled as a boolean.


In Kamailio script, see this example:

if (!jansson_get("method", $rb, "$var(method)")) {
xhttp_reply("400", "Bad Request", "", "");
exit;
}

This code works, but I'm not sure why.
Command "jansson_get" returns 1 if successful, and -1 if fails. Both values 
should be false.
Which I can see when testing with this:

$var(r) = jansson_get("method", $rb, "$var(method)");
xlog("L_INFO", "jansson_get r: [$var(r)]\n");
if (!$var(r)) {
xlog("L_INFO", "jansson_get r: [$var(r)] => FALSE\n");
} else {
xlog("L_INFO", "jansson_get r: [$var(r)] => TRUE\n");
}

In all cases, the result is always TRUE.
Is there some kind of "magic" conversion when directly checking the return of a 
command in Kamailio script ?


With KEMI python, the equivalent does not work:

if not KSR.jansson.get("method", KSR.pv.get("$rb"), 
"$var(method)"):
KSR.xhttp.xhttp_reply(400, "Bad Request", "", "")
return 1

So I have to do the following instead:

if KSR.jansson.get("method", KSR.pv.get("$rb"), "$var(method)") 
< 0:
KSR.xhttp.xhttp_reply(400, "Bad Request", "", "")
return 1



Thank you for your help.

Regards,
Nicolas.

This message contains information that may be privileged or confidential and is 
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[SR-Users] Question about UAC and PUA modules routing behavior

[Jean-Francois Lemieux]

  Hello, I have some trouble to figure out how routing works when the
request coming from the kamailio itself (used as a user agent per example)
.I try to use presence PUA... modules. The issue I have is the NOTIFY
requests aren't sent to the good destination; it doesn't seems to obey to
the contact or other informations when the device did a SUBSCRIBE.

I don't want to ask to troubleshoot the presence module directly

I just want to know how Kamailio deals with request coming from Kamailio
itself (in the case of a NOTIFY generated by PUA_... modules


My mains questions are:

Does the pua modules when generating request  like NOTIFY will pass through
request_route ?

Does the pua modules when generating  request  like NOTIFY will be sent
only when t_relay() is used or UA kamailio based request bypass all the
logic of the Kamailio proxy behavior ?

I'm trying to understand if I can modify the NOTIFY request to change
destination before the "routing decision" is made by Kamailio


Kamailio 5.6.0 used as :
-proxy to remote pbx for call dialog
-registrar/location server
-media relaying and nat management
-...and I try to use it as a presence server instead to use the pbx behind
kamailio
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Re: [SR-Users] Question about the Secsipid module

[Володимир Іванець]

Hello Daniel,

Thank you for the hints. I made a very simple configuration and Kamailio
v5.6 started without a problem. I will be looking into my current
configuration.

Test config:

loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "pv.so"


loadmodule "secsipid.so"


request_route {


  if ($si == "127.0.0.1") {
$var(secsipid_x5u) = "https://xyz/123K";;
$var(secsipid_keyPath) =
"/var/kamailio/stir-shaken/rule-aaa/132.key.pem";
secsipid_add_identity("$fU", "$rU", "A", "", "$var(secsipid_x5u)",
"$var(secsipid_keyPath)");
  }


  else {
if (!secsipid_check_identity("")) {
  sl_send_reply("403", "Forbidden");
  exit;
}
  }


  t_relay();
}


вт, 27 вер. 2022 р. о 18:42 Daniel-Constantin Mierla 
пише:

> You can use the 5.6 branch, secsipid module should be the same as in
> master branch.
>
> Otherwise, C99 should be required to be supported by the compiler because
> of code in many other components, is rpm build script enforcing an older C
> standard in the compiler?
>
> Cheers,
> Daniel
> On 27.09.22 17:31, Володимир Іванець wrote:
>
> I will try the master branch too. Unfortunately there is a problem with
> the RPM build procedure. I will try a different approach and update you
> when I have something. Thank you for your help!
>
>
> In file included from core/ppcfg.c:37:0:
> core/utils/snexpr.h: In function 'snexpr_op':
> core/utils/snexpr.h:272:2: error: 'for' loop initial declarations are only
> allowed in C99 mode
>   for(unsigned int i = 0; i < sizeof(OPS) / sizeof(OPS[0]); i++) {
>   ^
> core/utils/snexpr.h:272:2: note: use option -std=c99 or -std=gnu99 to
> compile your code
> core/utils/snexpr.h: In function 'snexpr_parse_number':
> core/utils/snexpr.h:286:2: error: 'for' loop initial declarations are only
> allowed in C99 mode
>   for(unsigned int i = 0; i < len; i++) {
>   ^
> core/utils/snexpr.h: In function 'snexpr_func_find':
> core/utils/snexpr.h:322:2: error: 'for' loop initial declarations are only
> allowed in C99 mode
>   for(struct snexpr_func *f = funcs; f->name; f++) {
>   ^
> core/utils/snexpr.h: In function 'snexpr_convert_num':
> core/utils/snexpr.h:398:3: warning: implicit declaration of function
> 'asprintf' [-Wimplicit-function-declaration]
>asprintf(&e->param.stz.sval, "%g", value);
>^
> core/utils/snexpr.h: In function 'snexpr_create':
> core/utils/snexpr.h:1193:6: error: 'for' loop initial declarations are
> only allowed in C99 mode
>   for(struct snexpr_var *v = vars->head; v; v = v->next) {
>   ^
> core/utils/snexpr.h:1217:7: error: 'for' loop initial declarations are
> only allowed in C99 mode
>for(int j = 0; j < sne_vec_len(&arg.args); j++) {
>^
> core/utils/snexpr.h:1231:15: error: redefinition of 'j'
>for(int j = 1; j < sne_vec_len(&m.body); j++) {
>^
> core/utils/snexpr.h:1217:15: note: previous definition of 'j' was here
>for(int j = 0; j < sne_vec_len(&arg.args); j++) {
>^
> core/utils/snexpr.h:1231:7: error: 'for' loop initial declarations are
> only allowed in C99 mode
>for(int j = 1; j < sne_vec_len(&m.body); j++) {
>^
> core/utils/snexpr.h: In function 'snexpr_destroy':
> core/utils/snexpr.h:1412:3: error: 'for' loop initial declarations are
> only allowed in C99 mode
>for(struct snexpr_var *v = vars->head; v;) {
>^
> make[1]: Leaving directory `/builddir/build/BUILD/kamailio-5.7.0-dev1/src'
> make[1]: *** [core/ppcfg.o] Error 1
> make: *** [default] Error 2
> error: Bad exit status from /var/tmp/rpm-tmp.I4sU6o (%build)
> Bad exit status from /var/tmp/rpm-tmp.I4sU6o (%build)
>
>
>
> RPM build errors:
> ERROR: Exception(../../kamailio-5.7.0-dev1.0.el7.centos.src.rpm)
> Config(epel-7-x86_64) 1 minutes 45 seconds
> INFO: Results and/or logs in: /var/lib/mock/epel-7-x86_64/result
> ERROR: Command failed. See logs for output.
>  # bash --login -c /usr/bin/rpmbuild -bb --target x86_64 --nodeps
> /builddir/build/SPECS/kamailio.spec
> make: *** [rpm] Error 1
>
>
> вт, 27 вер. 2022 р. о 17:50 Daniel-Constantin Mierla 
> пише:
>
>> Hello,
>>
>> I just tested with git master branch using:
>>
>> secsipid_check_identity("");
>>
>> and started fine. Maybe is something else wrong in your config.
>>
>> Cheers,
>> Daniel
>> On 27.09.22 13:58, Володимир Іванець wrote:
>>
>> Hello Daniel,
>>
>> Sorry for the delayed response. I can confirm that I get the "unknown
>> command, missing loadmodule?" error when using an empty parameter *if
>> (secsipid_check_identity(""))*.
>>
>> And thank you for the clarification about the file with the certificate.
>>
>> пн, 26 вер. 2022 р. о 19:50 Daniel-Constantin Mierla 
>> пише:
>>
>>> Hello,
>>>
>>> secsipid_check_identity("") should be ok, because the parameter is
>>> empty, but secsipid_check_identity() is not ok, because is no parameter.
>>>
>>> Isn't it working when you have secsipid_check_identity("")?
>>>
>>> It has to be one certificate, which corresponds to the key that was used
>>> to 

Re: [SR-Users] Question about the Secsipid module

[Daniel-Constantin Mierla]

You can use the 5.6 branch, secsipid module should be the same as in
master branch.

Otherwise, C99 should be required to be supported by the compiler
because of code in many other components, is rpm build script enforcing
an older C standard in the compiler?

Cheers,
Daniel

On 27.09.22 17:31, Володимир Іванець wrote:
> I will try the master branch too. Unfortunately there is a problem
> with the RPM build procedure. I will try a different approach and
> update you when I have something. Thank you for your help!
>
>
> In file included from core/ppcfg.c:37:0:
> core/utils/snexpr.h: In function 'snexpr_op':
> core/utils/snexpr.h:272:2: error: 'for' loop initial declarations
> are only allowed in C99 mode
>   for(unsigned int i = 0; i < sizeof(OPS) / sizeof(OPS[0]); i++) {
>   ^
> core/utils/snexpr.h:272:2: note: use option -std=c99 or -std=gnu99
> to compile your code
> core/utils/snexpr.h: In function 'snexpr_parse_number':
> core/utils/snexpr.h:286:2: error: 'for' loop initial declarations
> are only allowed in C99 mode
>   for(unsigned int i = 0; i < len; i++) {
>   ^
> core/utils/snexpr.h: In function 'snexpr_func_find':
> core/utils/snexpr.h:322:2: error: 'for' loop initial declarations
> are only allowed in C99 mode
>   for(struct snexpr_func *f = funcs; f->name; f++) {
>   ^
> core/utils/snexpr.h: In function 'snexpr_convert_num':
> core/utils/snexpr.h:398:3: warning: implicit declaration of
> function 'asprintf' [-Wimplicit-function-declaration]
>    asprintf(&e->param.stz.sval, "%g", value);
>    ^
> core/utils/snexpr.h: In function 'snexpr_create':
> core/utils/snexpr.h:1193:6: error: 'for' loop initial declarations
> are only allowed in C99 mode
>       for(struct snexpr_var *v = vars->head; v; v = v->next) {
>       ^
> core/utils/snexpr.h:1217:7: error: 'for' loop initial declarations
> are only allowed in C99 mode
>        for(int j = 0; j < sne_vec_len(&arg.args); j++) {
>        ^
> core/utils/snexpr.h:1231:15: error: redefinition of 'j'
>        for(int j = 1; j < sne_vec_len(&m.body); j++) {
>                ^
> core/utils/snexpr.h:1217:15: note: previous definition of 'j' was here
>        for(int j = 0; j < sne_vec_len(&arg.args); j++) {
>                ^
> core/utils/snexpr.h:1231:7: error: 'for' loop initial declarations
> are only allowed in C99 mode
>        for(int j = 1; j < sne_vec_len(&m.body); j++) {
>        ^
> core/utils/snexpr.h: In function 'snexpr_destroy':
> core/utils/snexpr.h:1412:3: error: 'for' loop initial declarations
> are only allowed in C99 mode
>    for(struct snexpr_var *v = vars->head; v;) {
>    ^
> make[1]: Leaving directory
> `/builddir/build/BUILD/kamailio-5.7.0-dev1/src'
> make[1]: *** [core/ppcfg.o] Error 1
> make: *** [default] Error 2
> error: Bad exit status from /var/tmp/rpm-tmp.I4sU6o (%build)
>     Bad exit status from /var/tmp/rpm-tmp.I4sU6o (%build)
>
>
>
> RPM build errors:
> ERROR: Exception(../../kamailio-5.7.0-dev1.0.el7.centos.src.rpm)
> Config(epel-7-x86_64) 1 minutes 45 seconds
> INFO: Results and/or logs in: /var/lib/mock/epel-7-x86_64/result
> ERROR: Command failed. See logs for output.
>  # bash --login -c /usr/bin/rpmbuild -bb --target x86_64 --nodeps
> /builddir/build/SPECS/kamailio.spec
> make: *** [rpm] Error 1
>
>
> вт, 27 вер. 2022 р. о 17:50 Daniel-Constantin Mierla
>  пише:
>
> Hello,
>
> I just tested with git master branch using:
>
> secsipid_check_identity("");
>
> and started fine. Maybe is something else wrong in your config.
>
> Cheers,
> Daniel
>
> On 27.09.22 13:58, Володимир Іванець wrote:
>> Hello Daniel,
>>
>> Sorry for the delayed response. I can confirm that I get the
>> "unknown command, missing loadmodule?" error when using an empty
>> parameter /if (secsipid_check_identity(""))/.
>>
>> And thank you for the clarification about the file with the
>> certificate.
>>
>> пн, 26 вер. 2022 р. о 19:50 Daniel-Constantin Mierla
>>  пише:
>>
>> Hello,
>>
>> secsipid_check_identity("") should be ok, because the
>> parameter is empty, but secsipid_check_identity() is not ok,
>> because is no parameter.
>>
>> Isn't it working when you have secsipid_check_identity("")?
>>
>> It has to be one certificate, which corresponds to the key
>> that was used to generate the identity header. The function
>> does not expect many certificates and loop to see if one of
>> them corresponds to the key.
>>
>> Cheers,
>> Daniel
>>
>> On 26.09.22 17:53, Володимир Іванець wrote:
>>> Hello Daniel!
>>>
>>> I started with a simpler configuration: /if
>>> (secsipid_check_identity()) {/ and /if
>>> (secsipid_check_identity("")

Re: [SR-Users] Question about the Secsipid module

[Володимир Іванець]

I will try the master branch too. Unfortunately there is a problem with the
RPM build procedure. I will try a different approach and update you when I
have something. Thank you for your help!


In file included from core/ppcfg.c:37:0:
core/utils/snexpr.h: In function 'snexpr_op':
core/utils/snexpr.h:272:2: error: 'for' loop initial declarations are only
allowed in C99 mode
  for(unsigned int i = 0; i < sizeof(OPS) / sizeof(OPS[0]); i++) {
  ^
core/utils/snexpr.h:272:2: note: use option -std=c99 or -std=gnu99 to
compile your code
core/utils/snexpr.h: In function 'snexpr_parse_number':
core/utils/snexpr.h:286:2: error: 'for' loop initial declarations are only
allowed in C99 mode
  for(unsigned int i = 0; i < len; i++) {
  ^
core/utils/snexpr.h: In function 'snexpr_func_find':
core/utils/snexpr.h:322:2: error: 'for' loop initial declarations are only
allowed in C99 mode
  for(struct snexpr_func *f = funcs; f->name; f++) {
  ^
core/utils/snexpr.h: In function 'snexpr_convert_num':
core/utils/snexpr.h:398:3: warning: implicit declaration of function
'asprintf' [-Wimplicit-function-declaration]
   asprintf(&e->param.stz.sval, "%g", value);
   ^
core/utils/snexpr.h: In function 'snexpr_create':
core/utils/snexpr.h:1193:6: error: 'for' loop initial declarations are only
allowed in C99 mode
  for(struct snexpr_var *v = vars->head; v; v = v->next) {
  ^
core/utils/snexpr.h:1217:7: error: 'for' loop initial declarations are only
allowed in C99 mode
   for(int j = 0; j < sne_vec_len(&arg.args); j++) {
   ^
core/utils/snexpr.h:1231:15: error: redefinition of 'j'
   for(int j = 1; j < sne_vec_len(&m.body); j++) {
   ^
core/utils/snexpr.h:1217:15: note: previous definition of 'j' was here
   for(int j = 0; j < sne_vec_len(&arg.args); j++) {
   ^
core/utils/snexpr.h:1231:7: error: 'for' loop initial declarations are only
allowed in C99 mode
   for(int j = 1; j < sne_vec_len(&m.body); j++) {
   ^
core/utils/snexpr.h: In function 'snexpr_destroy':
core/utils/snexpr.h:1412:3: error: 'for' loop initial declarations are only
allowed in C99 mode
   for(struct snexpr_var *v = vars->head; v;) {
   ^
make[1]: Leaving directory `/builddir/build/BUILD/kamailio-5.7.0-dev1/src'
make[1]: *** [core/ppcfg.o] Error 1
make: *** [default] Error 2
error: Bad exit status from /var/tmp/rpm-tmp.I4sU6o (%build)
Bad exit status from /var/tmp/rpm-tmp.I4sU6o (%build)



RPM build errors:
ERROR: Exception(../../kamailio-5.7.0-dev1.0.el7.centos.src.rpm)
Config(epel-7-x86_64) 1 minutes 45 seconds
INFO: Results and/or logs in: /var/lib/mock/epel-7-x86_64/result
ERROR: Command failed. See logs for output.
 # bash --login -c /usr/bin/rpmbuild -bb --target x86_64 --nodeps
/builddir/build/SPECS/kamailio.spec
make: *** [rpm] Error 1


вт, 27 вер. 2022 р. о 17:50 Daniel-Constantin Mierla 
пише:

> Hello,
>
> I just tested with git master branch using:
>
> secsipid_check_identity("");
>
> and started fine. Maybe is something else wrong in your config.
>
> Cheers,
> Daniel
> On 27.09.22 13:58, Володимир Іванець wrote:
>
> Hello Daniel,
>
> Sorry for the delayed response. I can confirm that I get the "unknown
> command, missing loadmodule?" error when using an empty parameter *if
> (secsipid_check_identity(""))*.
>
> And thank you for the clarification about the file with the certificate.
>
> пн, 26 вер. 2022 р. о 19:50 Daniel-Constantin Mierla 
> пише:
>
>> Hello,
>>
>> secsipid_check_identity("") should be ok, because the parameter is empty,
>> but secsipid_check_identity() is not ok, because is no parameter.
>>
>> Isn't it working when you have secsipid_check_identity("")?
>>
>> It has to be one certificate, which corresponds to the key that was used
>> to generate the identity header. The function does not expect many
>> certificates and loop to see if one of them corresponds to the key.
>>
>> Cheers,
>> Daniel
>> On 26.09.22 17:53, Володимир Іванець wrote:
>>
>> Hello Daniel!
>>
>> I started with a simpler configuration: *if (secsipid_check_identity())
>> {* and *if (secsipid_check_identity("")) {* - according to
>> documentation: *If the parameter is empty, the function is downloading
>> the key using the URL from "info" parameter of the Identity header, using
>> the value od "timeout" parameter to limit the download time.* If this
>> matters, I did not configure modparam "expire" and "timeout".
>>
>> When I called secsipid_check_identity with a path to the certificate,
>> Kamailio started as expected.
>>
>> Is there something else I need to do or the function
>> *secsipid_check_identity* must receive a path to certificate as
>> parameter? Also, is it safe to assume that this file can contain multiple
>> certificates?
>>
>> Thank you very much!
>>
>> пн, 26 вер. 2022 р. о 17:36 Daniel-Constantin Mierla 
>> пише:
>>
>>> Hello,
>>>
>>> maybe the function is provided with invalid parameters? What's the line
>>> 47 in the file?
>>>
>>> Cheers,
>>> Daniel
>>> On 26.09.22 15:54, Володимир Іванець wrote

Re: [SR-Users] Question about the Secsipid module

[Daniel-Constantin Mierla]

Hello,

I just tested with git master branch using:

secsipid_check_identity("");

and started fine. Maybe is something else wrong in your config.

Cheers,
Daniel

On 27.09.22 13:58, Володимир Іванець wrote:
> Hello Daniel,
>
> Sorry for the delayed response. I can confirm that I get the "unknown
> command, missing loadmodule?" error when using an empty parameter /if
> (secsipid_check_identity(""))/.
>
> And thank you for the clarification about the file with the certificate.
>
> пн, 26 вер. 2022 р. о 19:50 Daniel-Constantin Mierla
>  пише:
>
> Hello,
>
> secsipid_check_identity("") should be ok, because the parameter is
> empty, but secsipid_check_identity() is not ok, because is no
> parameter.
>
> Isn't it working when you have secsipid_check_identity("")?
>
> It has to be one certificate, which corresponds to the key that
> was used to generate the identity header. The function does not
> expect many certificates and loop to see if one of them
> corresponds to the key.
>
> Cheers,
> Daniel
>
> On 26.09.22 17:53, Володимир Іванець wrote:
>> Hello Daniel!
>>
>> I started with a simpler configuration: /if
>> (secsipid_check_identity()) {/ and /if
>> (secsipid_check_identity("")) {/ - according to documentation:
>> /If the parameter is empty, the function is downloading the key
>> using the URL from "info" parameter of the Identity header, using
>> the value od "timeout" parameter to limit the download time./ If
>> this matters, I did not configure modparam "expire" and "timeout".
>>
>> When I called secsipid_check_identity with a path to the
>> certificate, Kamailio started as expected.
>>
>> Is there something else I need to do or the
>> function /secsipid_check_identity/ must receive a path to
>> certificate as parameter? Also, is it safe to assume that this
>> file can contain multiple certificates?
>>
>> Thank you very much!
>>
>> пн, 26 вер. 2022 р. о 17:36 Daniel-Constantin Mierla
>>  пише:
>>
>> Hello,
>>
>> maybe the function is provided with invalid parameters?
>> What's the line 47 in the file?
>>
>> Cheers,
>> Daniel
>>
>> On 26.09.22 15:54, Володимир Іванець wrote:
>>> Hello everyone!
>>>
>>> I'm testing calls with STIR/SHAKEN with a Secsipid module on
>>> a Kamailio version 5.6.1 and ran into a problem. Outbound
>>> calls work fine when the function /secsipid_add_identity/
>>> with appropriate parameters is called. But as soon as I add
>>> function /secsipid_check_identity/ to my configuration,
>>> Kamilio would not start. Here is an error message I get:
>>>
>>> /kamailio: CRITICAL:  [core/cfg.y:3791]:
>>> yyerror_at(): parse error in config file
>>> /etc/kamailio/include/registrar.cfg, line 47, column 39:
>>> unknown command, missing loadmodule?/
>>>
>>> /
>>> /
>>> /loadmodule "secsipid.so"/ is present and is above the
>>> function /secsipid_check_identity/ call. Both secsipid.so
>>> and secsipid_proc.so module files are the correct version
>>> and present on the system.
>>>
>>>
>>> Another question is related to the rpm package build. How do
>>> you build additional modules? I made adjustment to
>>> the pkg/kamailio/Makefile file. In the *cfg* section
>>> changed /$(MAKE) -C ../../src cfg/ with /$(MAKE)
>>> FLAVOUR=kamailio include_modules="secsipid secsipid_proc" -C
>>> ../../src cfg/. Required modules appeared in the modules.lst
>>> file but their rpm packages were not built.
>>>
>>> Thank you very much!
>>>
>>> __
>>> Kamailio - Users Mailing List - Non Commercial Discussions
>>>   * sr-users@lists.kamailio.org
>>> Important: keep the mailing list in the recipients, do not reply 
>>> only to the sender!
>>> Edit mailing list options or unsubscribe:
>>>   * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>> -- 
>> Daniel-Constantin Mierla -- www.asipto.com 
>> www.twitter.com/miconda  -- 
>> www.linkedin.com/in/miconda 
>> Kamailio Advanced Training - Online
>>   Nov 7-10, 2022 (Europe Timezone)
>>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>>
> -- 
> Daniel-Constantin Mierla -- www.asipto.com 
> www.twitter.com/miconda  -- 
> www.linkedin.com/in/miconda 
> Kamailio Advanced Training - Online
>   Nov 7-10, 2022 (Europe Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
-- 
Daniel-Constantin Mierla -

Re: [SR-Users] Question about the Secsipid module

[Володимир Іванець]

Hello Daniel,

Sorry for the delayed response. I can confirm that I get the "unknown
command, missing loadmodule?" error when using an empty parameter *if
(secsipid_check_identity(""))*.

And thank you for the clarification about the file with the certificate.

пн, 26 вер. 2022 р. о 19:50 Daniel-Constantin Mierla 
пише:

> Hello,
>
> secsipid_check_identity("") should be ok, because the parameter is empty,
> but secsipid_check_identity() is not ok, because is no parameter.
>
> Isn't it working when you have secsipid_check_identity("")?
>
> It has to be one certificate, which corresponds to the key that was used
> to generate the identity header. The function does not expect many
> certificates and loop to see if one of them corresponds to the key.
>
> Cheers,
> Daniel
> On 26.09.22 17:53, Володимир Іванець wrote:
>
> Hello Daniel!
>
> I started with a simpler configuration: *if (secsipid_check_identity()) {*
> and *if (secsipid_check_identity("")) {* - according to documentation: *If
> the parameter is empty, the function is downloading the key using the URL
> from "info" parameter of the Identity header, using the value od "timeout"
> parameter to limit the download time.* If this matters, I did not
> configure modparam "expire" and "timeout".
>
> When I called secsipid_check_identity with a path to the certificate,
> Kamailio started as expected.
>
> Is there something else I need to do or the function
> *secsipid_check_identity* must receive a path to certificate as
> parameter? Also, is it safe to assume that this file can contain multiple
> certificates?
>
> Thank you very much!
>
> пн, 26 вер. 2022 р. о 17:36 Daniel-Constantin Mierla 
> пише:
>
>> Hello,
>>
>> maybe the function is provided with invalid parameters? What's the line
>> 47 in the file?
>>
>> Cheers,
>> Daniel
>> On 26.09.22 15:54, Володимир Іванець wrote:
>>
>> Hello everyone!
>>
>> I'm testing calls with STIR/SHAKEN with a Secsipid module on a Kamailio
>> version 5.6.1 and ran into a problem. Outbound calls work fine when the
>> function *secsipid_add_identity* with appropriate parameters is called.
>> But as soon as I add function *secsipid_check_identity* to my
>> configuration, Kamilio would not start. Here is an error message I get:
>>
>> *kamailio: CRITICAL:  [core/cfg.y:3791]: yyerror_at(): parse error
>> in config file /etc/kamailio/include/registrar.cfg, line 47, column 39:
>> unknown command, missing loadmodule?*
>>
>>
>> *loadmodule "secsipid.so"* is present and is above the function
>> *secsipid_check_identity* call. Both secsipid.so and secsipid_proc.so
>> module files are the correct version and present on the system.
>>
>>
>> Another question is related to the rpm package build. How do you build
>> additional modules? I made adjustment to the pkg/kamailio/Makefile file. In
>> the *cfg* section changed *$(MAKE) -C ../../src cfg* with *$(MAKE)
>> FLAVOUR=kamailio include_modules="secsipid secsipid_proc" -C ../../src cfg*.
>> Required modules appeared in the modules.lst file but their rpm packages
>> were not built.
>>
>> Thank you very much!
>>
>> __
>> Kamailio - Users Mailing List - Non Commercial Discussions
>>   * sr-users@lists.kamailio.org
>> Important: keep the mailing list in the recipients, do not reply only to the 
>> sender!
>> Edit mailing list options or unsubscribe:
>>   * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>> --
>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
>> www.linkedin.com/in/miconda
>> Kamailio Advanced Training - Online
>>   Nov 7-10, 2022 (Europe Timezone)
>>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>>
>> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
> www.linkedin.com/in/miconda
> Kamailio Advanced Training - Online
>   Nov 7-10, 2022 (Europe Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
>
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Re: [SR-Users] Question about the Secsipid module

[Daniel-Constantin Mierla]

Hello,

secsipid_check_identity("") should be ok, because the parameter is
empty, but secsipid_check_identity() is not ok, because is no parameter.

Isn't it working when you have secsipid_check_identity("")?

It has to be one certificate, which corresponds to the key that was used
to generate the identity header. The function does not expect many
certificates and loop to see if one of them corresponds to the key.

Cheers,
Daniel

On 26.09.22 17:53, Володимир Іванець wrote:
> Hello Daniel!
>
> I started with a simpler configuration: /if
> (secsipid_check_identity()) {/ and /if (secsipid_check_identity(""))
> {/ - according to documentation: /If the parameter is empty, the
> function is downloading the key using the URL from "info" parameter of
> the Identity header, using the value od "timeout" parameter to limit
> the download time./ If this matters, I did not configure modparam
> "expire" and "timeout".
>
> When I called secsipid_check_identity with a path to the certificate,
> Kamailio started as expected.
>
> Is there something else I need to do or the
> function /secsipid_check_identity/ must receive a path to certificate
> as parameter? Also, is it safe to assume that this file can contain
> multiple certificates?
>
> Thank you very much!
>
> пн, 26 вер. 2022 р. о 17:36 Daniel-Constantin Mierla
>  пише:
>
> Hello,
>
> maybe the function is provided with invalid parameters? What's the
> line 47 in the file?
>
> Cheers,
> Daniel
>
> On 26.09.22 15:54, Володимир Іванець wrote:
>> Hello everyone!
>>
>> I'm testing calls with STIR/SHAKEN with a Secsipid module on a
>> Kamailio version 5.6.1 and ran into a problem. Outbound calls
>> work fine when the function /secsipid_add_identity/ with
>> appropriate parameters is called. But as soon as I add function
>> /secsipid_check_identity/ to my configuration, Kamilio would not
>> start. Here is an error message I get:
>>
>> /kamailio: CRITICAL:  [core/cfg.y:3791]: yyerror_at():
>> parse error in config file
>> /etc/kamailio/include/registrar.cfg, line 47, column 39:
>> unknown command, missing loadmodule?/
>>
>> /
>> /
>> /loadmodule "secsipid.so"/ is present and is above the
>> function /secsipid_check_identity/ call. Both secsipid.so
>> and secsipid_proc.so module files are the correct version and
>> present on the system.
>>
>>
>> Another question is related to the rpm package build. How do you
>> build additional modules? I made adjustment to
>> the pkg/kamailio/Makefile file. In the *cfg* section
>> changed /$(MAKE) -C ../../src cfg/ with /$(MAKE) FLAVOUR=kamailio
>> include_modules="secsipid secsipid_proc" -C ../../src cfg/.
>> Required modules appeared in the modules.lst file but their rpm
>> packages were not built.
>>
>> Thank you very much!
>>
>> __
>> Kamailio - Users Mailing List - Non Commercial Discussions
>>   * sr-users@lists.kamailio.org
>> Important: keep the mailing list in the recipients, do not reply only to 
>> the sender!
>> Edit mailing list options or unsubscribe:
>>   * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> -- 
> Daniel-Constantin Mierla -- www.asipto.com 
> www.twitter.com/miconda  -- 
> www.linkedin.com/in/miconda 
> Kamailio Advanced Training - Online
>   Nov 7-10, 2022 (Europe Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - Online
  Nov 7-10, 2022 (Europe Timezone)
  * https://www.asipto.com/sw/kamailio-advanced-training-online/
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Re: [SR-Users] Question about the Secsipid module

[Володимир Іванець]

Hello Daniel!

I started with a simpler configuration: *if (secsipid_check_identity()) {*
and *if (secsipid_check_identity("")) {* - according to documentation: *If
the parameter is empty, the function is downloading the key using the URL
from "info" parameter of the Identity header, using the value od "timeout"
parameter to limit the download time.* If this matters, I did not configure
modparam "expire" and "timeout".

When I called secsipid_check_identity with a path to the certificate,
Kamailio started as expected.

Is there something else I need to do or the function
*secsipid_check_identity* must receive a path to certificate as parameter?
Also, is it safe to assume that this file can contain multiple certificates?

Thank you very much!

пн, 26 вер. 2022 р. о 17:36 Daniel-Constantin Mierla 
пише:

> Hello,
>
> maybe the function is provided with invalid parameters? What's the line 47
> in the file?
>
> Cheers,
> Daniel
> On 26.09.22 15:54, Володимир Іванець wrote:
>
> Hello everyone!
>
> I'm testing calls with STIR/SHAKEN with a Secsipid module on a Kamailio
> version 5.6.1 and ran into a problem. Outbound calls work fine when the
> function *secsipid_add_identity* with appropriate parameters is called.
> But as soon as I add function *secsipid_check_identity* to my
> configuration, Kamilio would not start. Here is an error message I get:
>
> *kamailio: CRITICAL:  [core/cfg.y:3791]: yyerror_at(): parse error
> in config file /etc/kamailio/include/registrar.cfg, line 47, column 39:
> unknown command, missing loadmodule?*
>
>
> *loadmodule "secsipid.so"* is present and is above the function
> *secsipid_check_identity* call. Both secsipid.so and secsipid_proc.so
> module files are the correct version and present on the system.
>
>
> Another question is related to the rpm package build. How do you build
> additional modules? I made adjustment to the pkg/kamailio/Makefile file. In
> the *cfg* section changed *$(MAKE) -C ../../src cfg* with *$(MAKE)
> FLAVOUR=kamailio include_modules="secsipid secsipid_proc" -C ../../src cfg*.
> Required modules appeared in the modules.lst file but their rpm packages
> were not built.
>
> Thank you very much!
>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
>   * sr-users@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to the 
> sender!
> Edit mailing list options or unsubscribe:
>   * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
> www.linkedin.com/in/miconda
> Kamailio Advanced Training - Online
>   Nov 7-10, 2022 (Europe Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
>
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Re: [SR-Users] Question about the Secsipid module

[Daniel-Constantin Mierla]

Hello,

maybe the function is provided with invalid parameters? What's the line
47 in the file?

Cheers,
Daniel

On 26.09.22 15:54, Володимир Іванець wrote:
> Hello everyone!
>
> I'm testing calls with STIR/SHAKEN with a Secsipid module on a
> Kamailio version 5.6.1 and ran into a problem. Outbound calls work
> fine when the function /secsipid_add_identity/ with appropriate
> parameters is called. But as soon as I add function
> /secsipid_check_identity/ to my configuration, Kamilio would not
> start. Here is an error message I get:
>
> /kamailio: CRITICAL:  [core/cfg.y:3791]: yyerror_at(): parse
> error in config file /etc/kamailio/include/registrar.cfg, line 47,
> column 39: unknown command, missing loadmodule?/
>
> /
> /
> /loadmodule "secsipid.so"/ is present and is above the
> function /secsipid_check_identity/ call. Both secsipid.so
> and secsipid_proc.so module files are the correct version and present
> on the system.
>
>
> Another question is related to the rpm package build. How do you build
> additional modules? I made adjustment to the pkg/kamailio/Makefile
> file. In the *cfg* section changed /$(MAKE) -C ../../src cfg/
> with /$(MAKE) FLAVOUR=kamailio include_modules="secsipid
> secsipid_proc" -C ../../src cfg/. Required modules appeared in
> the modules.lst file but their rpm packages were not built.
>
> Thank you very much!
>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
>   * sr-users@lists.kamailio.org
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> sender!
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-- 
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www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - Online
  Nov 7-10, 2022 (Europe Timezone)
  * https://www.asipto.com/sw/kamailio-advanced-training-online/
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[SR-Users] Question about the Secsipid module

[Володимир Іванець]

Hello everyone!

I'm testing calls with STIR/SHAKEN with a Secsipid module on a Kamailio
version 5.6.1 and ran into a problem. Outbound calls work fine when the
function *secsipid_add_identity* with appropriate parameters is called. But
as soon as I add function *secsipid_check_identity* to my configuration,
Kamilio would not start. Here is an error message I get:

*kamailio: CRITICAL:  [core/cfg.y:3791]: yyerror_at(): parse error in
config file /etc/kamailio/include/registrar.cfg, line 47, column 39:
unknown command, missing loadmodule?*


*loadmodule "secsipid.so"* is present and is above the function
*secsipid_check_identity* call. Both secsipid.so and secsipid_proc.so
module files are the correct version and present on the system.


Another question is related to the rpm package build. How do you build
additional modules? I made adjustment to the pkg/kamailio/Makefile file. In
the *cfg* section changed *$(MAKE) -C ../../src cfg* with *$(MAKE)
FLAVOUR=kamailio include_modules="secsipid secsipid_proc" -C ../../src cfg*.
Required modules appeared in the modules.lst file but their rpm packages
were not built.

Thank you very much!
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[SR-Users] Question about SIP registration

[Cheng Tang]

Hi, everyone!

In one sentence, I want to know how to how to add the shared key (aka. K, Ki) 
used by AKA algorithm to Kamailio.

I’m using Kamailio as an IMS server to test my own 5G Core. As you know, UE 
would send its first SIP Registration message to Kamailio with its IMPI and 
IMPU, then Kamailio should use the shared key to proceed AKA algorithm to 
generate the authentication vector, then respond UE with 401 unauthorized 
message to challenge the UE. I’ve done my research and I still can’t figure out 
how to let Kamailio know this shared key. And when I was doing my test, it 
seems like Kamailio used some default key to generate the auth vector.

Could anybody help me? I’ve been stuck here for days.
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Re: [SR-Users] Question on pike_check_ip(ipaddr)

[Fred Posner]

On 5/13/22 16:38, Joel Serrano wrote:
I was wondering if pike_check_ip(ipaddr) needs strictly an IP as 
parameter or can it be a string?


pike_check_ip is when you want to use an IP other than the source ip...

You can always use an htable to store the ip's of your high volume 
clients...


something like:

if((src_ip!=myself) && ($sht(highvolcust=>$si) == $null)) {
if (!pike_check_req()) {
...
}
}

--fred

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[SR-Users] Question on pike_check_ip(ipaddr)

[Joel Serrano]

Hello,

One of our customers has lots of extensions that register to our platform
using all the same IP. Sometimes they get caught in our pike velocity rules
but if we increase the values to accommodate this client then all the
others could abuse the system.

I was wondering if pike_check_ip(ipaddr) needs strictly an IP as
parameter or can it be a string?

I was thinking of using pike instead of by $si by something like $fU-$si
but I don't know if this would work?

I need to either handle separately different extensions from the same IP or
a way to adjust the pike config values just for one specific IP... I'm not
sure either of these is possible with pike?

My alternative is just use table and mimic the functionality myself which I
don't think is too complicated, but I'd love to know if I'm missing a
fast-path here...

Thanks!
Joel.
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Re: [SR-Users] Question about a future 5.6 release

[Daniel-Constantin Mierla]

Hello,


it's getting close for a new major release, indeed. Probably we should
freeze development by end of this month, test during April and release
sometime in May.


Cheers,
Daniel


On 16.03.22 09:39, Stefan-Cristian Mititelu wrote:
>
> Hi,
>
> I want to ask you if there is an approximate timeline for a kamailio
> 5.6 release?
>
> We are planning an upgrade on our systems and are weighing different
> options.
>
>  
> Thank you
> Stefan
>
>
> __
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[SR-Users] Question about a future 5.6 release

[Stefan-Cristian Mititelu]

Hi,

I want to ask you if there is an approximate timeline for a kamailio 5.6 
release?

We are planning an upgrade on our systems and are weighing different options.


Thank you
Stefan
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Re: [SR-Users] Question about siptrace destination/source port capture in HEPv3 mode

[Sergey Safarov]

you can check the ticket
https://github.com/kamailio/kamailio/issues/2092

On Thu, Jan 20, 2022 at 3:21 PM Daniel-Constantin Mierla 
wrote:

> Hello,
>
> the source port of tcp connections is typically an ephemeral port, that's
> due to tcp design and os layer implementation. Newer kernels have the
> option for reuse port for tcp, but even then it cannot be guaranteed always
> -- there should be an option for recent kamailios as well.
>
> Maybe it can be enforced inside the c code of siptrace modue, when the hep
> header is built. I expect port 0 is set when it is not known for whatsoever
> reason.
>
> Cheers,
> Daniel
> On 18.01.22 13:13, Karsten Horsmann wrote:
>
> Hello List,
>
> I observe an interesting behavior. We use siptrace as an active capture
> agent for QXIP HOMER/HEPIC installations.
> The destination port and the source port in the HEPv3 Header are not
> correct if you use transport tcp for example.
>
> My Question: is that intended or a bug and i should fill up an issue?
>
> On udp, it's filled with 5060 or whatever your listening directive says.
>
> I tested that with kamailio 5.5 and kamailio master.
>
> I attached an HEPv3 capture from two kamailios (master) speaking with each
> other.
> You need the https://github.com/sipcapture/hep-wireshark lua dissector to
> read the capture.
>
> 10.0.2.15 (kamailio with uacreg and sipcapture/HEPv3 mode)
> 192.168.50.4 (kamailio as registrar)
>
> Frame 1 shows an tcp transport register Request from 10.0.2.15 to
> 192.168.50.4 in HEPv3.
> The Source Port in the HEP3 Protocol is not correct (its an highport /
> 45419).
>
> Frame 4 shows the 401 answer and the Destination port is 0.
> Frame 7 also had this behavior.
>
>
> - params (nearly at the end):
> modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE")
> modparam("siptrace", "hep_mode_on", 1)
> modparam("siptrace", "hep_version", 3)
> modparam("siptrace", "hep_capture_id", MY_HOMER_CAPTURE_ID) # capture
> agent id Limitation: 32-bit for HEPv3.
> modparam("siptrace", "trace_to_database", 0)
> modparam("siptrace", "trace_on", 1)
> modparam("siptrace", "trace_mode", 1) # default 0, if 1 then you dont need
> call siptrace flag or sip_trace()
>
>
>
> --
> Mit freundlichen Grüßen / Kind Regards
> *Karsten Horsmann*
>
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> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
> www.linkedin.com/in/miconda
> Kamailio Advanced Training - Online
>   Feb 21-24, 2022 (America Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
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Re: [SR-Users] Question about siptrace destination/source port capture in HEPv3 mode

[Daniel-Constantin Mierla]

Hello,

the source port of tcp connections is typically an ephemeral port,
that's due to tcp design and os layer implementation. Newer kernels have
the option for reuse port for tcp, but even then it cannot be guaranteed
always -- there should be an option for recent kamailios as well.

Maybe it can be enforced inside the c code of siptrace modue, when the
hep header is built. I expect port 0 is set when it is not known for
whatsoever reason.

Cheers,
Daniel

On 18.01.22 13:13, Karsten Horsmann wrote:
> Hello List,
>
> I observe an interesting behavior. We use siptrace as an active
> capture agent for QXIP HOMER/HEPIC installations.
> The destination port and the source port in the HEPv3 Header are not
> correct if you use transport tcp for example.
>
> My Question: is that intended or a bug and i should fill up an issue?
>
> On udp, it's filled with 5060 or whatever your listening directive says.
>
> I tested that with kamailio 5.5 and kamailio master.
>
> I attached an HEPv3 capture from two kamailios (master) speaking with
> each other.
> You need the https://github.com/sipcapture/hep-wireshark lua dissector
> to read the capture.
>
> 10.0.2.15 (kamailio with uacreg and sipcapture/HEPv3 mode)
> 192.168.50.4 (kamailio as registrar)
>
> Frame 1 shows an tcp transport register Request from 10.0.2.15 to
> 192.168.50.4 in HEPv3.
> The Source Port in the HEP3 Protocol is not correct (its an highport /
> 45419).
>
> Frame 4 shows the 401 answer and the Destination port is 0.
> Frame 7 also had this behavior.
>
>
> - params (nearly at the end):
> modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE")
> modparam("siptrace", "hep_mode_on", 1)
> modparam("siptrace", "hep_version", 3)
> modparam("siptrace", "hep_capture_id", MY_HOMER_CAPTURE_ID) # capture
> agent id Limitation: 32-bit for HEPv3.
> modparam("siptrace", "trace_to_database", 0)
> modparam("siptrace", "trace_on", 1)
> modparam("siptrace", "trace_mode", 1) # default 0, if 1 then you dont
> need call siptrace flag or sip_trace()
>
>
>
> -- 
> Mit freundlichen Grüßen / Kind Regards
> *Karsten Horsmann*
>
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> sender!
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-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - Online
  Feb 21-24, 2022 (America Timezone)
  * https://www.asipto.com/sw/kamailio-advanced-training-online/
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[SR-Users] Question about siptrace destination/source port capture in HEPv3 mode

[Karsten Horsmann]

Hello List,

I observe an interesting behavior. We use siptrace as an active capture
agent for QXIP HOMER/HEPIC installations.
The destination port and the source port in the HEPv3 Header are not
correct if you use transport tcp for example.

My Question: is that intended or a bug and i should fill up an issue?

On udp, it's filled with 5060 or whatever your listening directive says.

I tested that with kamailio 5.5 and kamailio master.

I attached an HEPv3 capture from two kamailios (master) speaking with each
other.
You need the https://github.com/sipcapture/hep-wireshark lua dissector to
read the capture.

10.0.2.15 (kamailio with uacreg and sipcapture/HEPv3 mode)
192.168.50.4 (kamailio as registrar)

Frame 1 shows an tcp transport register Request from 10.0.2.15 to
192.168.50.4 in HEPv3.
The Source Port in the HEP3 Protocol is not correct (its an highport /
45419).

Frame 4 shows the 401 answer and the Destination port is 0.
Frame 7 also had this behavior.


- params (nearly at the end):
modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE")
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "hep_version", 3)
modparam("siptrace", "hep_capture_id", MY_HOMER_CAPTURE_ID) # capture agent
id Limitation: 32-bit for HEPv3.
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "trace_mode", 1) # default 0, if 1 then you dont need
call siptrace flag or sip_trace()



-- 
Mit freundlichen Grüßen / Kind Regards
*Karsten Horsmann*


dumphep3.pcap
Description: Binary data
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Re: [SR-Users] Question regarding parser error for REGISTER request

[Henning Westerholt]

Hello,

the parser error could be also generated from some unrelated operation in the 
cfg file, like accessing the Contact parameters. You should be able to get from 
the context of the error log more information.
Regarding debugging, for an obvious suggestion - have you already tried to just 
dump the complete SIP message in the main route for debugging, e.g., with INFO 
log level?

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>

From: sr-users  On Behalf Of Floimair 
Florian
Sent: Monday, September 20, 2021 2:19 PM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Question regarding parser error for REGISTER request

Hello List!

We have a certain user agent that somehow triggers a parser error log entry, 
everytime this type of user agent registers.
Sep 20 12:03:49 kamailio-dev-1 /usr/sbin/kamailio[1803]: ERROR:  
[core/parser/parse_param.c:517]: parse_param2(): Invalid character, ; expected

The odd thing now is that this register still succeeds. And we do not see that 
error for other methods like e.g. INVITE.
The user agent in question is an Android app based on linphone which we develop 
inhouse. We also use the same for iOS where the error does not occur.

The REGISTER request contains a push-token in the Contact header and I think 
this is where the parser somehow generates this error. Since this is the only 
line apart from the User-Agent header where the content of the two is different.
We did have a look and diffed the differences between our iOS and Android app 
variants and didn’t see any obvious problem or invalid character (compared even 
with hex editor).

So my questions regarding this I hope that someone of you can help me with:

-  Is there any way to make the parser error more verbose (e.g. which 
character in which header it is complaining about)? Right now it only says that 
there was an error and that it expected ‘;’ but not where and what character 
was found instead.

-  Why does the parser throw the error but the registration still 
succeeds? It does not quite make sense to me.


Best Regards

FLORIAN FLOIMAIR
Development
Commend International GmbH
Saalachstrasse 51
5020 Salzburg, Austria
Phone: +43 662 85 62 25
Mail: f.floim...@commend.com<mailto:f.floim...@commend.com>
[signature_1687123261]
commend.com
LG Salzburg / FN 178618z

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Re: [SR-Users] Question on the SL Module

[Henning Westerholt]

Hello,

I agree that this stats handling is a bit non-intuitive.

Basically, the reasoning behind the implementation was, to count certain, 
well-known return codes with individual counters and have a generic NXX counter 
for the rest.

Taking the 5xx codes as example, the 500 counter just gives you this value, and 
the 5XX counter the rest. If you want to get all 5xx codes counts, you need to 
take the sum of 500 and 5XX counter. Similar logic also applies to the other 
counters in this module.

Hope this makes it clear. Extensions to the docs are of course always welcome, 
just open a pull-request.

Cheers,

Henning

-- 
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://gilawa.com 

-Original Message-
From: sr-users  On Behalf Of Alessio Garzi
Sent: Monday, September 20, 2021 5:10 PM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Question on the SL Module

Hello Kama ML,

today I stumbled on function "update_sl_stats()" inside sl_stats.c and noticed 
that some response codes are treated differently.

For example here:

https://github.com/kamailio/kamailio/blob/a9ef78b7f43616f2ceaa26df97eff68eb5348846/src/modules/sl/sl_stats.c#L165


since we have a "break;" statement, when a 500 code is received, the 5XX 
counter is not incremented even if a 500 code belongs to the 5XX family.


We have the same behaviour for other response codes and it really seems 
that it's not a bug and the author really wanted it.

The docs are not so helpful and they don't explain much in detail how 
counters are built.

I just want to have confirmation this is how it must work and I wonder 
if it's the case to write something about it into the docs.

Thanks

Alessio



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[SR-Users] Question on the SL Module

[Alessio Garzi]

Hello Kama ML,

today I stumbled on function "update_sl_stats()" inside sl_stats.c and 
noticed that some response codes are treated differently.


For example here:

https://github.com/kamailio/kamailio/blob/a9ef78b7f43616f2ceaa26df97eff68eb5348846/src/modules/sl/sl_stats.c#L165


since we have a "break;" statement, when a 500 code is received, the 5XX 
counter is not incremented even if a 500 code belongs to the 5XX family.



We have the same behaviour for other response codes and it really seems 
that it's not a bug and the author really wanted it.


The docs are not so helpful and they don't explain much in detail how 
counters are built.


I just want to have confirmation this is how it must work and I wonder 
if it's the case to write something about it into the docs.


Thanks

Alessio



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[SR-Users] Question regarding parser error for REGISTER request

[Floimair Florian]

Hello List!

We have a certain user agent that somehow triggers a parser error log entry, 
everytime this type of user agent registers.

Sep 20 12:03:49 kamailio-dev-1 /usr/sbin/kamailio[1803]: ERROR:  
[core/parser/parse_param.c:517]: parse_param2(): Invalid character, ; expected

The odd thing now is that this register still succeeds. And we do not see that 
error for other methods like e.g. INVITE.
The user agent in question is an Android app based on linphone which we develop 
inhouse. We also use the same for iOS where the error does not occur.

The REGISTER request contains a push-token in the Contact header and I think 
this is where the parser somehow generates this error. Since this is the only 
line apart from the User-Agent header where the content of the two is different.
We did have a look and diffed the differences between our iOS and Android app 
variants and didn’t see any obvious problem or invalid character (compared even 
with hex editor).

So my questions regarding this I hope that someone of you can help me with:


  *   Is there any way to make the parser error more verbose (e.g. which 
character in which header it is complaining about)? Right now it only says that 
there was an error and that it expected ‘;’ but not where and what character 
was found instead.
  *   Why does the parser throw the error but the registration still succeeds? 
It does not quite make sense to me.


Best Regards

FLORIAN FLOIMAIR
Development
Commend International GmbH
Saalachstrasse 51
5020 Salzburg, Austria
Phone: +43 662 85 62 25
Mail: f.floim...@commend.com
[signature_1687123261]
commend.com
LG Salzburg / FN 178618z

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Re: [SR-Users] Question about suspected memory leak presence module

[Henning Westerholt]

Hello Albert,

4.4.7 is over three years old. Before you invest more time in this issue, I 
would suggest that you update to a maintained version (e.g. 5.4.x or 5.5.x).
There have been many extensions to the code base and also bug fixes to since 
this time.

The numbers from the memory debugging below basically list the allocated memory 
from these functions. They should normally not grow endless.

There are also ways to output information to the log file, described e.g. here: 
http://www.kamailio.org/wiki/tutorials/troubleshooting/memory

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>

From: Albert Petit 
Sent: Monday, September 13, 2021 10:34 PM
To: Henning Westerholt 
Cc: Kamailio (SER) - Users Mailing List ; SIP 
Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List 

Subject: Re: [SR-Users] Question about suspected memory leak presence module

hi

using kamailio 4.4.7 on that environment

i will try to run some test locally simulating reception of a lot presence 
subscription and  observe mem_copy_subs_noc(),
Just to understand,  both mem_copy_subs_noc are not expected to increase, is 
that right? What do they exactly mean?

Thanks


On Sat, 11 Sep 2021, 22:22 Henning Westerholt, 
mailto:h...@skalatan.de>> wrote:
Hello,

the output below indicates the memory allocated in the individual functions. So 
it seems that the mem_copy_subs_noc() function allocates a lot of memory. It is 
increasing over time? The final output looks like a wrapping error, the counter 
goes negative.

Which version of Kamailio do you use on the


server?

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Albert Petit
Sent: Thursday, September 9, 2021 11:42 PM
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users 
Mailing List 
mailto:sr-us...@lists.sip-router.org>>
Subject: [SR-Users] Question about suspected memory leak presence module

Hi,

I am suspecitng presence module might be having a memory leak on a production 
server used by an end user (not have access to it).

The output of command kamcmd mod.stats presence shm is
Module: presence
{
mem_copy_subs(148): 25720
mem_copy_subs_noc(214): 1672201904
mem_copy_subs_noc(251): 678406560
add_event(181): 88
shm_copy_event(57): 40
shm_copy_event(50): 312
add_event(156): 40
add_event(149): 456
new_shtable(66): 5767168
new_shtable(53): 262144
init_evlist(289): 16
Total: -1938302848
}

I am not sure what mem_copy_subs_noc is meaning. And final output is negative,  
SHM is incfreasing all the time but not sure of the root cause

is this a reasonable hint that a memory leak is related to presence?
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Re: [SR-Users] Question about suspected memory leak presence module

[Albert Petit]

hi

using kamailio 4.4.7 on that environment

i will try to run some test locally simulating reception of a lot presence
subscription and  observe mem_copy_subs_noc(),
Just to understand,  both mem_copy_subs_noc are not expected to increase,
is that right? What do they exactly mean?

Thanks


On Sat, 11 Sep 2021, 22:22 Henning Westerholt,  wrote:

> Hello,
>
>
>
> the output below indicates the memory allocated in the individual
> functions. So it seems that the mem_copy_subs_noc() function allocates a
> lot of memory. It is increasing over time? The final output looks like a
> wrapping error, the counter goes negative.
>
>
>
> Which version of Kamailio do you use on the
>


server?
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* sr-users  *On Behalf Of *Albert
> Petit
> *Sent:* Thursday, September 9, 2021 11:42 PM
> *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List 
> *Subject:* [SR-Users] Question about suspected memory leak presence module
>
>
>
> Hi,
>
>
>
> I am suspecitng presence module might be having a memory leak on a
> production server used by an end user (not have access to it).
>
>
>
> The output of command kamcmd mod.stats presence shm is
>
> Module: presence
>
> {
>
> mem_copy_subs(148): 25720
>
> mem_copy_subs_noc(214): 1672201904
>
> mem_copy_subs_noc(251): 678406560
>
> add_event(181): 88
>
> shm_copy_event(57): 40
>
> shm_copy_event(50): 312
>
> add_event(156): 40
>
> add_event(149): 456
>
> new_shtable(66): 5767168
>
> new_shtable(53): 262144
>
> init_evlist(289): 16
>
> Total: -1938302848
>
> }
>
>
>
> I am not sure what mem_copy_subs_noc is meaning. And final output is
> negative,  SHM is incfreasing all the time but not sure of the root cause
>
>
>
> is this a reasonable hint that a memory leak is related to presence?
>
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Re: [SR-Users] Question about suspected memory leak presence module

[Henning Westerholt]

Hello,

the output below indicates the memory allocated in the individual functions. So 
it seems that the mem_copy_subs_noc() function allocates a lot of memory. It is 
increasing over time? The final output looks like a wrapping error, the counter 
goes negative.

Which version of Kamailio do you use on the server?

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>

From: sr-users  On Behalf Of Albert Petit
Sent: Thursday, September 9, 2021 11:42 PM
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users 
Mailing List 
Subject: [SR-Users] Question about suspected memory leak presence module

Hi,

I am suspecitng presence module might be having a memory leak on a production 
server used by an end user (not have access to it).

The output of command kamcmd mod.stats presence shm is
Module: presence
{
mem_copy_subs(148): 25720
mem_copy_subs_noc(214): 1672201904
mem_copy_subs_noc(251): 678406560
add_event(181): 88
shm_copy_event(57): 40
shm_copy_event(50): 312
add_event(156): 40
add_event(149): 456
new_shtable(66): 5767168
new_shtable(53): 262144
init_evlist(289): 16
Total: -1938302848
}

I am not sure what mem_copy_subs_noc is meaning. And final output is negative,  
SHM is incfreasing all the time but not sure of the root cause

is this a reasonable hint that a memory leak is related to presence?
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[SR-Users] Question about suspected memory leak presence module

[Albert Petit]

Hi,

I am suspecitng presence module might be having a memory leak on a
production server used by an end user (not have access to it).

The output of command kamcmd mod.stats presence shm is

Module: presence

{

mem_copy_subs(148): 25720

mem_copy_subs_noc(214): 1672201904

mem_copy_subs_noc(251): 678406560

add_event(181): 88

shm_copy_event(57): 40

shm_copy_event(50): 312

add_event(156): 40

add_event(149): 456

new_shtable(66): 5767168

new_shtable(53): 262144

init_evlist(289): 16

Total: -1938302848

}

I am not sure what mem_copy_subs_noc is meaning. And final output is
negative,  SHM is incfreasing all the time but not sure of the root cause

is this a reasonable hint that a memory leak is related to presence?
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Re: [SR-Users] Question about MS Teams integration

[Володимир Іванець]

I figured this out. Hope this will help someone as it was not obvious to me
at first.

My test user was assigned to the "mydomain.onmicrosoft.com" domain. I
guess this is what "tenant" means in the "Make sure the domain name portion
of the FQDN matches a domain *that's registered in your tenant* ..."
sentence from the documentation. As soon as I changed it to "mydomain.com"
I was able to create SBC with a "kamailio-dev2.mydomain.com" FQDN.

Thanks for the help!

Regards, Volodymyr Ivanets.

чт, 22 лип. 2021 о 14:34 Володимир Іванець  пише:

> Hello Rob,
>
> Thank you for sharing this information!
>
> Regards, Volodymyr Ivanets.
>
> ср, 21 лип. 2021 о 21:12  пише:
>
>>
>>
>> *Van:* sr-users  *Namens *?
>> ???
>> *Verzonden:* woensdag 21 juli 2021 19:54
>> *Aan:* Kamailio (SER) - Users Mailing List 
>> *Onderwerp:* Re: [SR-Users] Question about MS Teams integration
>>
>>
>>
>> Rob,
>>
>>
>>
>> Thank you for the link!
>>
>>
>>
>> My domain record looks good in the Settings/Domains section. However, I
>> only choose the "Skype for Business" option. I did not select "Exchange and
>> Exchange Online Protection" as well as "Intune and Mobile Device Management
>> for Microsoft 365" options.
>>
>>
>>
>> According to the tutorial I need to use PowerShell to configure SBC,
>> Direct routing and call routing rules. Is this really necessary? I could
>> create a dummy SBC for the "mydomain.onmicrosoft.com" domain with UI.
>>
>>
>>
>>
>>
>> Hello Vitalie!
>>
>>
>>
>> Could you please tell me if the Settings/Domains is the only place where
>> the domain has to be configured? And do I need to have a specific list of
>> records? Currently, I only have CNAME records "sip" and "lyncdiscover"
>> snd SRV records "_sip" and "_sipfederationtls".
>>
>>
>>
>>
>>
>> Thank you all for the help!
>>
>>
>>
>> Regards, Volodymyr Ivanets.
>>
>>
>>
>> ср, 21 лип. 2021 о 20:03 Bugaian A. Vitalie  пише:
>>
>> Hi Volodimir,
>>
>>
>>
>> you have to setup and verify your domain under admin.microsoft.com in
>> domains section,then create users under that domain
>>
>> and make a dialing plan for these users so you can call these users
>> from/to  kamailio.
>>
>>
>>
>> Vitalie.
>>
>>
>>
>> On Wed, Jul 21, 2021 at 6:21 PM Володимир Іванець <
>> volodyaivan...@gmail.com> wrote:
>>
>> Hello all!
>>
>>
>>
>> Sorry, this is not a direct Kamailio question. I would like to ask for
>> some help from those who already did this.
>>
>>
>>
>> I'm trying to integrate my testKamailio installation with MS Teams. I
>> added the "mydomain.com" to the Domains section (I kept the "
>> mydomain.onmicrosoft.com"). When I tried to create SBC with a "
>> kamailio-dev2.mydomain.com" FQDN I got this error message: We* can't use
>> the "kamailio-dev2.mydomain.com <http://kamailio-dev2.mydomain.com>" domain
>> as it hasn't be set up in the organization. Try again. If you continue to
>> have problems, contact Microsoft customer support.*
>>
>>
>>
>> There were no issues creating a dummy SBC with a "
>> kamailio-dev2.mydomain.onmicrosoft.com" FQDN.
>>
>>
>>
>> Does anyone know how to set up a domain in the organization from MS
>> perspective?
>>
>>
>>
>> Thank you very much!
>>
>>
>>
>> Regards, Volodymyr Ivanets.
>>
>> __
>> Kamailio - Users Mailing List - Non Commercial Discussions
>>   * sr-users@lists.kamailio.org
>> Important: keep the mailing list in the recipients, do not reply only to
>> the sender!
>> Edit mailing list options or unsubscribe:
>>   * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>> __
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>>   * sr-users@lists.kamailio.org
>> Important: keep the mailing list in the recipients, do not reply only to
>> the sender!
>> Edit mailing list options or unsubscribe:
>>   * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>> __
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>> Important: keep the mailing list in the recipients, do not reply only to
>> the sender!
>> Edit mailing list options or unsubscribe:
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>>
>
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Re: [SR-Users] Question about MS Teams integration

[Володимир Іванець]

Hello Rob,

Thank you for sharing this information!

Regards, Volodymyr Ivanets.

ср, 21 лип. 2021 о 21:12  пише:

>
>
> *Van:* sr-users  *Namens *?
> ???
> *Verzonden:* woensdag 21 juli 2021 19:54
> *Aan:* Kamailio (SER) - Users Mailing List 
> *Onderwerp:* Re: [SR-Users] Question about MS Teams integration
>
>
>
> Rob,
>
>
>
> Thank you for the link!
>
>
>
> My domain record looks good in the Settings/Domains section. However, I
> only choose the "Skype for Business" option. I did not select "Exchange and
> Exchange Online Protection" as well as "Intune and Mobile Device Management
> for Microsoft 365" options.
>
>
>
> According to the tutorial I need to use PowerShell to configure SBC,
> Direct routing and call routing rules. Is this really necessary? I could
> create a dummy SBC for the "mydomain.onmicrosoft.com" domain with UI.
>
>
>
>
>
> Hello Vitalie!
>
>
>
> Could you please tell me if the Settings/Domains is the only place where
> the domain has to be configured? And do I need to have a specific list of
> records? Currently, I only have CNAME records "sip" and "lyncdiscover"
> snd SRV records "_sip" and "_sipfederationtls".
>
>
>
>
>
> Thank you all for the help!
>
>
>
> Regards, Volodymyr Ivanets.
>
>
>
> ср, 21 лип. 2021 о 20:03 Bugaian A. Vitalie  пише:
>
> Hi Volodimir,
>
>
>
> you have to setup and verify your domain under admin.microsoft.com in
> domains section,then create users under that domain
>
> and make a dialing plan for these users so you can call these users
> from/to  kamailio.
>
>
>
> Vitalie.
>
>
>
> On Wed, Jul 21, 2021 at 6:21 PM Володимир Іванець <
> volodyaivan...@gmail.com> wrote:
>
> Hello all!
>
>
>
> Sorry, this is not a direct Kamailio question. I would like to ask for
> some help from those who already did this.
>
>
>
> I'm trying to integrate my testKamailio installation with MS Teams. I
> added the "mydomain.com" to the Domains section (I kept the "
> mydomain.onmicrosoft.com"). When I tried to create SBC with a "
> kamailio-dev2.mydomain.com" FQDN I got this error message: We* can't use
> the "kamailio-dev2.mydomain.com <http://kamailio-dev2.mydomain.com>" domain
> as it hasn't be set up in the organization. Try again. If you continue to
> have problems, contact Microsoft customer support.*
>
>
>
> There were no issues creating a dummy SBC with a "
> kamailio-dev2.mydomain.onmicrosoft.com" FQDN.
>
>
>
> Does anyone know how to set up a domain in the organization from MS
> perspective?
>
>
>
> Thank you very much!
>
>
>
> Regards, Volodymyr Ivanets.
>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
>   * sr-users@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> Edit mailing list options or unsubscribe:
>   * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
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Re: [SR-Users] Question about MS Teams integration

[rob.van.den.bulk]

 

Van: sr-users  Namens ? ???
Verzonden: woensdag 21 juli 2021 19:54
Aan: Kamailio (SER) - Users Mailing List 
Onderwerp: Re: [SR-Users] Question about MS Teams integration

 

Rob,

 

Thank you for the link!

 

My domain record looks good in the Settings/Domains section. However, I only 
choose the "Skype for Business" option. I did not select "Exchange and Exchange 
Online Protection" as well as "Intune and Mobile Device Management for 
Microsoft 365" options.

 

According to the tutorial I need to use PowerShell to configure SBC, Direct 
routing and call routing rules. Is this really necessary? I could create a 
dummy SBC for the "mydomain.onmicrosoft.com <http://mydomain.onmicrosoft.com> " 
domain with UI.

 

 

Hello Vitalie!

 

Could you please tell me if the Settings/Domains is the only place where the 
domain has to be configured? And do I need to have a specific list of records? 
Currently, I only have CNAME records "sip" and "lyncdiscover" snd SRV records 
"_sip" and "_sipfederationtls".

 

 

Thank you all for the help!

 

Regards, Volodymyr Ivanets.

 

ср, 21 лип. 2021 о 20:03 Bugaian A. Vitalie mailto:buga...@gmail.com> > пише:

Hi Volodimir,

 

you have to setup and verify your domain under admin.microsoft.com 
<http://admin.microsoft.com/>  in domains section,then create users under that 
domain

and make a dialing plan for these users so you can call these users from/to  
kamailio.

 

Vitalie.

 

On Wed, Jul 21, 2021 at 6:21 PM Володимир Іванець mailto:volodyaivan...@gmail.com> > wrote:

Hello all!

 

Sorry, this is not a direct Kamailio question. I would like to ask for some 
help from those who already did this.

 

I'm trying to integrate my testKamailio installation with MS Teams. I added the 
"mydomain.com <http://mydomain.com> " to the Domains section (I kept the 
"mydomain.onmicrosoft.com <http://mydomain.onmicrosoft.com> "). When I tried to 
create SBC with a "kamailio-dev2.mydomain.com 
<http://kamailio-dev2.mydomain.com> " FQDN I got this error message: We can't 
use the "kamailio-dev2.mydomain.com <http://kamailio-dev2.mydomain.com> " 
domain as it hasn't be set up in the organization. Try again. If you continue 
to have problems, contact Microsoft customer support.

 

There were no issues creating a dummy SBC with a 
"kamailio-dev2.mydomain.onmicrosoft.com 
<http://kamailio-dev2.mydomain.onmicrosoft.com> " FQDN.

 

Does anyone know how to set up a domain in the organization from MS perspective?

 

Thank you very much!

 

Regards, Volodymyr Ivanets.

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Re: [SR-Users] Question about MS Teams integration

[Володимир Іванець]

Rob,

Thank you for the link!

My domain record looks good in the Settings/Domains section. However, I
only choose the "Skype for Business" option. I did not select "Exchange and
Exchange Online Protection" as well as "Intune and Mobile Device Management
for Microsoft 365" options.

According to the tutorial I need to use PowerShell to configure SBC, Direct
routing and call routing rules. Is this really necessary? I could create a
dummy SBC for the "mydomain.onmicrosoft.com" domain with UI.


Hello Vitalie!

Could you please tell me if the Settings/Domains is the only place where
the domain has to be configured? And do I need to have a specific list of
records? Currently, I only have CNAME records "sip" and "lyncdiscover" snd
SRV records "_sip" and "_sipfederationtls".


Thank you all for the help!

Regards, Volodymyr Ivanets.

ср, 21 лип. 2021 о 20:03 Bugaian A. Vitalie  пише:

> Hi Volodimir,
>
> you have to setup and verify your domain under admin.microsoft.com in
> domains section,then create users under that domain
> and make a dialing plan for these users so you can call these users
> from/to  kamailio.
>
> Vitalie.
>
> On Wed, Jul 21, 2021 at 6:21 PM Володимир Іванець <
> volodyaivan...@gmail.com> wrote:
>
>> Hello all!
>>
>> Sorry, this is not a direct Kamailio question. I would like to ask for
>> some help from those who already did this.
>>
>> I'm trying to integrate my testKamailio installation with MS Teams. I
>> added the "mydomain.com" to the Domains section (I kept the "
>> mydomain.onmicrosoft.com"). When I tried to create SBC with a "
>> kamailio-dev2.mydomain.com" FQDN I got this error message: We* can't use
>> the "kamailio-dev2.mydomain.com " domain
>> as it hasn't be set up in the organization. Try again. If you continue to
>> have problems, contact Microsoft customer support.*
>>
>> There were no issues creating a dummy SBC with a "
>> kamailio-dev2.mydomain.onmicrosoft.com" FQDN.
>>
>> Does anyone know how to set up a domain in the organization from MS
>> perspective?
>>
>> Thank you very much!
>>
>> Regards, Volodymyr Ivanets.
>> __
>> Kamailio - Users Mailing List - Non Commercial Discussions
>>   * sr-users@lists.kamailio.org
>> Important: keep the mailing list in the recipients, do not reply only to
>> the sender!
>> Edit mailing list options or unsubscribe:
>>   * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
> __
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>   * sr-users@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
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Re: [SR-Users] Question about MS Teams integration

[Bugaian A. Vitalie]

Hi Volodimir,

you have to setup and verify your domain under admin.microsoft.com in
domains section,then create users under that domain
and make a dialing plan for these users so you can call these users
from/to  kamailio.

Vitalie.

On Wed, Jul 21, 2021 at 6:21 PM Володимир Іванець 
wrote:

> Hello all!
>
> Sorry, this is not a direct Kamailio question. I would like to ask for
> some help from those who already did this.
>
> I'm trying to integrate my testKamailio installation with MS Teams. I
> added the "mydomain.com" to the Domains section (I kept the "
> mydomain.onmicrosoft.com"). When I tried to create SBC with a "
> kamailio-dev2.mydomain.com" FQDN I got this error message: We* can't use
> the "kamailio-dev2.mydomain.com " domain
> as it hasn't be set up in the organization. Try again. If you continue to
> have problems, contact Microsoft customer support.*
>
> There were no issues creating a dummy SBC with a "
> kamailio-dev2.mydomain.onmicrosoft.com" FQDN.
>
> Does anyone know how to set up a domain in the organization from MS
> perspective?
>
> Thank you very much!
>
> Regards, Volodymyr Ivanets.
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
>   * sr-users@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> Edit mailing list options or unsubscribe:
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Re: [SR-Users] Question about MS Teams integration

[rob.van.den.bulk]

Or this is better, 

 

https://www.vitalpbx.org/blog/integrate-vitalpbx-with-microsoft-teams/

 

The first part you can use for kamailio

 

 

 

Van: sr-users  Namens ? ???
Verzonden: woensdag 21 juli 2021 17:56
Aan: Kamailio (SER) - Users Mailing List 
Onderwerp: Re: [SR-Users] Question about MS Teams integration

 

Hello Rob!

 

I have to confess that I don't have any experience with Teams :)

 

I've added "mydomain.com <http://mydomain.com/> " to the Settings => Domains 
section in the Microsoft 365 admin center. Is Tenant a different thing? If yes 
could you please point me where to find it?

 

Thank you!

 

Regards, Volodymyr Ivanets.

 

ср, 21 лип. 2021 о 18:26 Rob van den Bulk mailto:rob.van.den.b...@gmail.com> > пише:

Dit you setup the domain in the tenantadmin.microsoft.com 
<http://admin.microsoft.com> 

 

Outlook voor Android <https://aka.ms/AAb9ysg>  downloaden

  _  

From: sr-users mailto:sr-users-boun...@lists.kamailio.org> > on behalf of Володимир Іванець 
mailto:volodyaivan...@gmail.com> >
Sent: Wednesday, July 21, 2021 5:20:16 PM
To: Kamailio (SER) - Users Mailing List mailto:sr-users@lists.kamailio.org> >
Subject: [SR-Users] Question about MS Teams integration 

 

Hello all! 

 

Sorry, this is not a direct Kamailio question. I would like to ask for some 
help from those who already did this.

 

I'm trying to integrate my testKamailio installation with MS Teams. I added the 
"mydomain.com <http://mydomain.com> " to the Domains section (I kept the 
"mydomain.onmicrosoft.com <http://mydomain.onmicrosoft.com> "). When I tried to 
create SBC with a "kamailio-dev2.mydomain.com 
<http://kamailio-dev2.mydomain.com> " FQDN I got this error message: We can't 
use the "kamailio-dev2.mydomain.com <http://kamailio-dev2.mydomain.com> " 
domain as it hasn't be set up in the organization. Try again. If you continue 
to have problems, contact Microsoft customer support.

 

There were no issues creating a dummy SBC with a 
"kamailio-dev2.mydomain.onmicrosoft.com 
<http://kamailio-dev2.mydomain.onmicrosoft.com> " FQDN.

 

Does anyone know how to set up a domain in the organization from MS perspective?

 

Thank you very much!

 

Regards, Volodymyr Ivanets.

__
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Re: [SR-Users] Question about MS Teams integration

[Rob van den Bulk]

I make you a small doc, I send it today, ist point you in the direction...



Outlook voor Android<https://aka.ms/AAb9ysg> downloaden

From: sr-users  on behalf of Володимир 
Іванець 
Sent: Wednesday, July 21, 2021 5:56:06 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Question about MS Teams integration

Hello Rob!

I have to confess that I don't have any experience with Teams :)

I've added "mydomain.com<http://mydomain.com/>" to the Settings => Domains 
section in the Microsoft 365 admin center. Is Tenant a different thing? If yes 
could you please point me where to find it?

Thank you!

Regards, Volodymyr Ivanets.

ср, 21 лип. 2021 о 18:26 Rob van den Bulk 
mailto:rob.van.den.b...@gmail.com>> пише:
Dit you setup the domain in the tenant
admin.microsoft.com<http://admin.microsoft.com>

Outlook voor Android<https://aka.ms/AAb9ysg> downloaden

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 on behalf of Володимир Іванець 
mailto:volodyaivan...@gmail.com>>
Sent: Wednesday, July 21, 2021 5:20:16 PM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Question about MS Teams integration

Hello all!

Sorry, this is not a direct Kamailio question. I would like to ask for some 
help from those who already did this.

I'm trying to integrate my testKamailio installation with MS Teams. I added the 
"mydomain.com<http://mydomain.com>" to the Domains section (I kept the 
"mydomain.onmicrosoft.com<http://mydomain.onmicrosoft.com>"). When I tried to 
create SBC with a 
"kamailio-dev2.mydomain.com<http://kamailio-dev2.mydomain.com>" FQDN I got this 
error message: We can't use the 
"kamailio-dev2.mydomain.com<http://kamailio-dev2.mydomain.com>" domain as it 
hasn't be set up in the organization. Try again. If you continue to have 
problems, contact Microsoft customer support.

There were no issues creating a dummy SBC with a 
"kamailio-dev2.mydomain.onmicrosoft.com<http://kamailio-dev2.mydomain.onmicrosoft.com>"
 FQDN.

Does anyone know how to set up a domain in the organization from MS perspective?

Thank you very much!

Regards, Volodymyr Ivanets.
__
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  * sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>
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Re: [SR-Users] Question about MS Teams integration

[Володимир Іванець]

Hello Rob!

I have to confess that I don't have any experience with Teams :)

I've added "mydomain.com" to the Settings => Domains section in the
Microsoft 365 admin center. Is Tenant a different thing? If yes could
you please point me where to find it?

Thank you!

Regards, Volodymyr Ivanets.

ср, 21 лип. 2021 о 18:26 Rob van den Bulk  пише:

> Dit you setup the domain in the tenantadmin.microsoft.com
>
> Outlook voor Android <https://aka.ms/AAb9ysg> downloaden
> --
> *From:* sr-users  on behalf of
> Володимир Іванець 
> *Sent:* Wednesday, July 21, 2021 5:20:16 PM
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* [SR-Users] Question about MS Teams integration
>
> Hello all!
>
> Sorry, this is not a direct Kamailio question. I would like to ask for
> some help from those who already did this.
>
> I'm trying to integrate my testKamailio installation with MS Teams. I
> added the "mydomain.com" to the Domains section (I kept the "
> mydomain.onmicrosoft.com"). When I tried to create SBC with a "
> kamailio-dev2.mydomain.com" FQDN I got this error message: We* can't use
> the "kamailio-dev2.mydomain.com <http://kamailio-dev2.mydomain.com>" domain
> as it hasn't be set up in the organization. Try again. If you continue to
> have problems, contact Microsoft customer support.*
>
> There were no issues creating a dummy SBC with a "
> kamailio-dev2.mydomain.onmicrosoft.com" FQDN.
>
> Does anyone know how to set up a domain in the organization from MS
> perspective?
>
> Thank you very much!
>
> Regards, Volodymyr Ivanets.
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
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Re: [SR-Users] Question about MS Teams integration

[Rob van den Bulk]

Dit you setup the domain in the tenantadmin.microsoft.com

Outlook voor Android<https://aka.ms/AAb9ysg> downloaden

From: sr-users  on behalf of Володимир 
Іванець 
Sent: Wednesday, July 21, 2021 5:20:16 PM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Question about MS Teams integration

Hello all!

Sorry, this is not a direct Kamailio question. I would like to ask for some 
help from those who already did this.

I'm trying to integrate my testKamailio installation with MS Teams. I added the 
"mydomain.com<http://mydomain.com>" to the Domains section (I kept the 
"mydomain.onmicrosoft.com<http://mydomain.onmicrosoft.com>"). When I tried to 
create SBC with a 
"kamailio-dev2.mydomain.com<http://kamailio-dev2.mydomain.com>" FQDN I got this 
error message: We can't use the 
"kamailio-dev2.mydomain.com<http://kamailio-dev2.mydomain.com>" domain as it 
hasn't be set up in the organization. Try again. If you continue to have 
problems, contact Microsoft customer support.

There were no issues creating a dummy SBC with a 
"kamailio-dev2.mydomain.onmicrosoft.com<http://kamailio-dev2.mydomain.onmicrosoft.com>"
 FQDN.

Does anyone know how to set up a domain in the organization from MS perspective?

Thank you very much!

Regards, Volodymyr Ivanets.
__
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[SR-Users] Question about MS Teams integration

[Володимир Іванець]

Hello all!

Sorry, this is not a direct Kamailio question. I would like to ask for
some help from those who already did this.

I'm trying to integrate my testKamailio installation with MS Teams. I added
the "mydomain.com" to the Domains section (I kept the "
mydomain.onmicrosoft.com"). When I tried to create SBC with a "
kamailio-dev2.mydomain.com" FQDN I got this error message: We* can't use
the "kamailio-dev2.mydomain.com " domain
as it hasn't be set up in the organization. Try again. If you continue to
have problems, contact Microsoft customer support.*

There were no issues creating a dummy SBC with a "
kamailio-dev2.mydomain.onmicrosoft.com" FQDN.

Does anyone know how to set up a domain in the organization from MS
perspective?

Thank you very much!

Regards, Volodymyr Ivanets.
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Re: [SR-Users] Question about Kamailio IMS in box

[Mojtaba]

absolutely yes.
Just set local DNS ip server in nslookup file in Ubuntu.
With regards.

On Wed, 12 May 2021, 20:48 Ibrahim soumana djibo, 
wrote:

> Hello,
>
> I am new to kamailio. And I Am following the work of Franz Edler.
>
> I would like to test it in a local environment without internet connection
> on ubuntu. What I don't understand is, to download the packages I
> necessarily need the internet. Do I have to use two network interfaces for
> the same virtual machine in order to access the internet and make the DNS
> configuration locally with the second interface?
>
> Cordially,
>
>
>
> *SOUMANA** DJIBO **Ibrahim*
> *Ingénieur Réseaux & Télécoms*
> *5 Avenue Emile Baudot, 91300 Massy*
> +33 07 61 80 20 35
> [image: Résultat de recherche d'images pour "logo paris saclay"]
> https://www.universite-paris-saclay.fr/
>
> __
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[SR-Users] Question about Kamailio IMS in box

[Ibrahim soumana djibo]

Hello,

I am new to kamailio. And I Am following the work of Franz Edler.

I would like to test it in a local environment without internet connection
on ubuntu. What I don't understand is, to download the packages I
necessarily need the internet. Do I have to use two network interfaces for
the same virtual machine in order to access the internet and make the DNS
configuration locally with the second interface?

Cordially,



*SOUMANA** DJIBO **Ibrahim*
*Ingénieur Réseaux & Télécoms*
*5 Avenue Emile Baudot, 91300 Massy*
+33 07 61 80 20 35
[image: Résultat de recherche d'images pour "logo paris saclay"]
https://www.universite-paris-saclay.fr/
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Re: [SR-Users] [Question] About softphone for VoLTE

[Daniel-Constantin Mierla]

Hello,

thanks for all these useful details!

Cheers,
Daniel

On 18.03.21 07:02, supreeth herle wrote:
> Hi Daniel,
>
> I am not using any developer mode for the phone app. I have tested
> using a test network with programmable SIM cards from Sysmocom and
> COTS UE. In order for the Android phones to perform an IMS attach I
> make use of Carrier Privileges to force enable the VoLTE feature. More
> instructions can be found in the following link
> https://github.com/herlesupreeth/CoIMS_Wiki
> . From my experience if
> you are using a test network (PLMN 00101) you have maximum success in
> activation of VoLTE on the phone.
>
> On the iPhones, the IMS is by default enabled for all PLMNs
>
> Best Regards,
> Supreeth
>
>
> On Wed, 17 Mar 2021 at 20:21, Daniel-Constantin Mierla
> mailto:mico...@gmail.com>> wrote:
>
> Hello,
>
> is there like a developer mode for this app that allows to
> register on a local LTE network? Or you tested with a simcard
> built for you own network?
>
> Cheers,
> Daniel
>
> On 17.03.21 15:05, supreeth herle wrote:
>> Hi Shigeru,
>>
>> Having written Open5GS (LTE / EPC) + Kamailio's VoLTE tutorial
>> myself here is my 2 cents on VoLTE client. You can use any
>> Android phone / iphone with its default phone app to test VoLTE
>> (try to avoid Samsung phone as much as possible)
>>
>> Best regards,
>> Supreeth
>>
>>
>> On Wed, Mar 17, 2021, 3:45 PM shigeru ishida
>> mailto:s5u.ish...@gmail.com>> wrote:
>>
>> Hi Daniel,
>>
>> Thank you very much for your information.
>>
>> I have tried AGEphone (VoIP) before.
>>
>> However, I don't have any information on whether Open5GS (LTE
>> / EPC) + Kamailio's VoLTE can be used with a normal phone app
>> on comercial smartphones.
>> Maybe it's just because I'm not familiar with this area.
>>
>> So, I asked if there is a general VoLTE compatible softphone.
>>
>> Thank you very much again.
>>
>> Regards,
>>
>> --Shigeru
>>
>> 2021年3月17日(水) 16:21 Daniel-Constantin Mierla
>> mailto:mico...@gmail.com>>:
>>
>> Hello,
>>
>> not acting much in this space, but I heard about
>> something like imsdroid -- probably you can try to search
>> on the net and see what you can find.
>>
>> Cheers,
>> Daniel
>>
>> On 12.03.21 09:01, shigeru ishida wrote:
>>>
>>> Hi all,
>>>
>>> I have one question.
>>>
>>> What softphones can I use when trying VoLTE (not VoIP)
>>> with Kamailio? I would like to verify VoLTE of private
>>> LTE with Open5GS and Kamailio.
>>>
>>> I'm sorry to ask you a question without much
>>> understanding of Kamailio, VoIP and VoLTE.
>>>
>>> Thank you very much in advance.
>>>
>>> Best regards,
>>>
>>> --Shigeru
>>>
>>>
>>> ___
>>> Kamailio (SER) - Users Mailing List
>>> sr-users@lists.kamailio.org 
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
>>> 
>>
>> -- 
>> Daniel-Constantin Mierla -- www.asipto.com 
>> 
>> www.twitter.com/miconda  -- 
>> www.linkedin.com/in/miconda 
>> Funding: https://www.paypal.me/dcmierla 
>> 
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> 
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>> 
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org 
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>> 
>>
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org 
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
>> 
>
> -- 
> Daniel-Constantin Mierla -- www.asipto.com 
> www.twitter.com/miconda  -- 
> www.linkedin.com/in/miconda 
> Funding: https://www.paypal.me/dcmierla 

Re: [SR-Users] [Question] About softphone for VoLTE

[supreeth herle]

Hi Daniel,

I am not using any developer mode for the phone app. I have tested using a
test network with programmable SIM cards from Sysmocom and COTS UE. In
order for the Android phones to perform an IMS attach I make use of Carrier
Privileges to force enable the VoLTE feature. More instructions can be
found in the following link https://github.com/herlesupreeth/CoIMS_Wiki.
>From my experience if you are using a test network (PLMN 00101) you have
maximum success in activation of VoLTE on the phone.

On the iPhones, the IMS is by default enabled for all PLMNs

Best Regards,
Supreeth


On Wed, 17 Mar 2021 at 20:21, Daniel-Constantin Mierla 
wrote:

> Hello,
>
> is there like a developer mode for this app that allows to register on a
> local LTE network? Or you tested with a simcard built for you own network?
>
> Cheers,
> Daniel
> On 17.03.21 15:05, supreeth herle wrote:
>
> Hi Shigeru,
>
> Having written Open5GS (LTE / EPC) + Kamailio's VoLTE tutorial myself
> here is my 2 cents on VoLTE client. You can use any Android phone / iphone
> with its default phone app to test VoLTE (try to avoid Samsung phone as
> much as possible)
>
> Best regards,
> Supreeth
>
>
> On Wed, Mar 17, 2021, 3:45 PM shigeru ishida  wrote:
>
>> Hi Daniel,
>>
>> Thank you very much for your information.
>>
>> I have tried AGEphone (VoIP) before.
>>
>> However, I don't have any information on whether Open5GS (LTE / EPC) +
>> Kamailio's VoLTE can be used with a normal phone app on comercial
>> smartphones.
>> Maybe it's just because I'm not familiar with this area.
>>
>> So, I asked if there is a general VoLTE compatible softphone.
>>
>> Thank you very much again.
>>
>> Regards,
>>
>> --Shigeru
>>
>> 2021年3月17日(水) 16:21 Daniel-Constantin Mierla :
>>
>>> Hello,
>>>
>>> not acting much in this space, but I heard about something like imsdroid
>>> -- probably you can try to search on the net and see what you can find.
>>>
>>> Cheers,
>>> Daniel
>>> On 12.03.21 09:01, shigeru ishida wrote:
>>>
>>> Hi all,
>>>
>>> I have one question.
>>>
>>> What softphones can I use when trying VoLTE (not VoIP) with Kamailio? I
>>> would like to verify VoLTE of private LTE with Open5GS and Kamailio.
>>>
>>> I'm sorry to ask you a question without much understanding of Kamailio,
>>> VoIP and VoLTE.
>>>
>>> Thank you very much in advance.
>>>
>>> Best regards,
>>>
>>> --Shigeru
>>>
>>> ___
>>> Kamailio (SER) - Users Mailing 
>>> Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>> --
>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
>>> www.linkedin.com/in/miconda
>>> Funding: https://www.paypal.me/dcmierla
>>>
>>> ___
>>> Kamailio (SER) - Users Mailing List
>>> sr-users@lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
> ___
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> Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
> www.linkedin.com/in/miconda
> Funding: https://www.paypal.me/dcmierla
>
>
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Re: [SR-Users] [Question] About softphone for VoLTE

[shigeru ishida]

Hi Supreeth,

Thank you very much for your information!

Is your URL as below?

https://github.com/herlesupreeth/Kamailio_IMS_Config

Regarding VoLTE, there is another URL on Open5GS site.

https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/

In any case, it seems that VoLTE built on private LTE is able to be used
with a default phone app of smartphones, so I will try it later.
(It seems to me to be a complicated setting :-)

Best regards,

--Shigeru

2021年3月17日(水) 23:07 supreeth herle :

> Hi Shigeru,
>
> Having written Open5GS (LTE / EPC) + Kamailio's VoLTE tutorial myself
> here is my 2 cents on VoLTE client. You can use any Android phone / iphone
> with its default phone app to test VoLTE (try to avoid Samsung phone as
> much as possible)
>
> Best regards,
> Supreeth
>
>
> On Wed, Mar 17, 2021, 3:45 PM shigeru ishida  wrote:
>
>> Hi Daniel,
>>
>> Thank you very much for your information.
>>
>> I have tried AGEphone (VoIP) before.
>>
>> However, I don't have any information on whether Open5GS (LTE / EPC) +
>> Kamailio's VoLTE can be used with a normal phone app on comercial
>> smartphones.
>> Maybe it's just because I'm not familiar with this area.
>>
>> So, I asked if there is a general VoLTE compatible softphone.
>>
>> Thank you very much again.
>>
>> Regards,
>>
>> --Shigeru
>>
>> 2021年3月17日(水) 16:21 Daniel-Constantin Mierla :
>>
>>> Hello,
>>>
>>> not acting much in this space, but I heard about something like imsdroid
>>> -- probably you can try to search on the net and see what you can find.
>>>
>>> Cheers,
>>> Daniel
>>> On 12.03.21 09:01, shigeru ishida wrote:
>>>
>>> Hi all,
>>>
>>> I have one question.
>>>
>>> What softphones can I use when trying VoLTE (not VoIP) with Kamailio? I
>>> would like to verify VoLTE of private LTE with Open5GS and Kamailio.
>>>
>>> I'm sorry to ask you a question without much understanding of Kamailio,
>>> VoIP and VoLTE.
>>>
>>> Thank you very much in advance.
>>>
>>> Best regards,
>>>
>>> --Shigeru
>>>
>>> ___
>>> Kamailio (SER) - Users Mailing 
>>> Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>> --
>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
>>> www.linkedin.com/in/miconda
>>> Funding: https://www.paypal.me/dcmierla
>>>
>>> ___
>>> Kamailio (SER) - Users Mailing List
>>> sr-users@lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
> ___
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> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
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Re: [SR-Users] [Question] About softphone for VoLTE

[Daniel-Constantin Mierla]

Hello,

is there like a developer mode for this app that allows to register on a
local LTE network? Or you tested with a simcard built for you own network?

Cheers,
Daniel

On 17.03.21 15:05, supreeth herle wrote:
> Hi Shigeru,
>
> Having written Open5GS (LTE / EPC) + Kamailio's VoLTE tutorial myself
> here is my 2 cents on VoLTE client. You can use any Android phone /
> iphone with its default phone app to test VoLTE (try to avoid Samsung
> phone as much as possible)
>
> Best regards,
> Supreeth
>
>
> On Wed, Mar 17, 2021, 3:45 PM shigeru ishida  > wrote:
>
> Hi Daniel,
>
> Thank you very much for your information.
>
> I have tried AGEphone (VoIP) before.
>
> However, I don't have any information on whether Open5GS (LTE /
> EPC) + Kamailio's VoLTE can be used with a normal phone app on
> comercial smartphones.
> Maybe it's just because I'm not familiar with this area.
>
> So, I asked if there is a general VoLTE compatible softphone.
>
> Thank you very much again.
>
> Regards,
>
> --Shigeru
>
> 2021年3月17日(水) 16:21 Daniel-Constantin Mierla  >:
>
> Hello,
>
> not acting much in this space, but I heard about something
> like imsdroid -- probably you can try to search on the net and
> see what you can find.
>
> Cheers,
> Daniel
>
> On 12.03.21 09:01, shigeru ishida wrote:
>>
>> Hi all,
>>
>> I have one question.
>>
>> What softphones can I use when trying VoLTE (not VoIP) with
>> Kamailio? I would like to verify VoLTE of private LTE with
>> Open5GS and Kamailio.
>>
>> I'm sorry to ask you a question without much understanding of
>> Kamailio, VoIP and VoLTE.
>>
>> Thank you very much in advance.
>>
>> Best regards,
>>
>> --Shigeru
>>
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org 
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
>> 
>
> -- 
> Daniel-Constantin Mierla -- www.asipto.com 
> www.twitter.com/miconda  -- 
> www.linkedin.com/in/miconda 
> Funding: https://www.paypal.me/dcmierla 
> 
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org 
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> 
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org 
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> 
>
>
> ___
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> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla

___
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Re: [SR-Users] [Question] About softphone for VoLTE

[supreeth herle]

Hi Shigeru,

Having written Open5GS (LTE / EPC) + Kamailio's VoLTE tutorial myself here
is my 2 cents on VoLTE client. You can use any Android phone / iphone with
its default phone app to test VoLTE (try to avoid Samsung phone as much as
possible)

Best regards,
Supreeth


On Wed, Mar 17, 2021, 3:45 PM shigeru ishida  wrote:

> Hi Daniel,
>
> Thank you very much for your information.
>
> I have tried AGEphone (VoIP) before.
>
> However, I don't have any information on whether Open5GS (LTE / EPC) +
> Kamailio's VoLTE can be used with a normal phone app on comercial
> smartphones.
> Maybe it's just because I'm not familiar with this area.
>
> So, I asked if there is a general VoLTE compatible softphone.
>
> Thank you very much again.
>
> Regards,
>
> --Shigeru
>
> 2021年3月17日(水) 16:21 Daniel-Constantin Mierla :
>
>> Hello,
>>
>> not acting much in this space, but I heard about something like imsdroid
>> -- probably you can try to search on the net and see what you can find.
>>
>> Cheers,
>> Daniel
>> On 12.03.21 09:01, shigeru ishida wrote:
>>
>> Hi all,
>>
>> I have one question.
>>
>> What softphones can I use when trying VoLTE (not VoIP) with Kamailio? I
>> would like to verify VoLTE of private LTE with Open5GS and Kamailio.
>>
>> I'm sorry to ask you a question without much understanding of Kamailio,
>> VoIP and VoLTE.
>>
>> Thank you very much in advance.
>>
>> Best regards,
>>
>> --Shigeru
>>
>> ___
>> Kamailio (SER) - Users Mailing 
>> Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>> --
>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
>> www.linkedin.com/in/miconda
>> Funding: https://www.paypal.me/dcmierla
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
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Re: [SR-Users] [Question] About softphone for VoLTE

[shigeru ishida]

Hi Daniel,

Thank you very much for your information.

I have tried AGEphone (VoIP) before.

However, I don't have any information on whether Open5GS (LTE / EPC) +
Kamailio's VoLTE can be used with a normal phone app on comercial
smartphones.
Maybe it's just because I'm not familiar with this area.

So, I asked if there is a general VoLTE compatible softphone.

Thank you very much again.

Regards,

--Shigeru

2021年3月17日(水) 16:21 Daniel-Constantin Mierla :

> Hello,
>
> not acting much in this space, but I heard about something like imsdroid
> -- probably you can try to search on the net and see what you can find.
>
> Cheers,
> Daniel
> On 12.03.21 09:01, shigeru ishida wrote:
>
> Hi all,
>
> I have one question.
>
> What softphones can I use when trying VoLTE (not VoIP) with Kamailio? I
> would like to verify VoLTE of private LTE with Open5GS and Kamailio.
>
> I'm sorry to ask you a question without much understanding of Kamailio,
> VoIP and VoLTE.
>
> Thank you very much in advance.
>
> Best regards,
>
> --Shigeru
>
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Re: [SR-Users] [Question] About softphone for VoLTE

[Daniel-Constantin Mierla]

Hello,

not acting much in this space, but I heard about something like imsdroid
-- probably you can try to search on the net and see what you can find.

Cheers,
Daniel

On 12.03.21 09:01, shigeru ishida wrote:
>
> Hi all,
>
> I have one question.
>
> What softphones can I use when trying VoLTE (not VoIP) with Kamailio?
> I would like to verify VoLTE of private LTE with Open5GS and Kamailio.
>
> I'm sorry to ask you a question without much understanding of
> Kamailio, VoIP and VoLTE.
>
> Thank you very much in advance.
>
> Best regards,
>
> --Shigeru
>
>
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[SR-Users] [Question] About softphone for VoLTE

[shigeru ishida]

Hi all,

I have one question.

What softphones can I use when trying VoLTE (not VoIP) with Kamailio? I
would like to verify VoLTE of private LTE with Open5GS and Kamailio.

I'm sorry to ask you a question without much understanding of Kamailio,
VoIP and VoLTE.

Thank you very much in advance.

Best regards,

--Shigeru
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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

Ok..that's interesting to know.  Thanks for your input Richard.  This helps
a lot.

On Fri, Dec 4, 2020 at 2:18 PM Richard Fuchs  wrote:

> On 04/12/2020 14.04, Andrew Chen wrote:
> > Sure...I understand ICE has its own setup workflow than SIP but it's
> > also important that rtpengine uses the rtp path that's negotiated in
> > the SIP or else it can cause confusion (to those who don't understand
> > ICE very well like me).
>
> There is no RTP path negotiated through SIP. There is only a number of
> possible paths, which are compiled from the list of candidates that each
> side advertises. ICE then checks which paths work, and then finally
> chooses the one with the highest priority.
>
> The only way to have a single media path that is negotiated through SIP
> is if you don't use ICE.
>
> Cheers
>
>
>
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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

Sure...I understand ICE has its own setup workflow than SIP but it's also
important that rtpengine uses the rtp path that's negotiated in the SIP or
else it can cause confusion (to those who don't understand ICE very well
like me).

On Fri, Dec 4, 2020 at 1:47 PM Richard Fuchs  wrote:

> On 04/12/2020 13.36, Andrew Chen wrote:
> > Hmm..that's interesting.  You would guess that the rtpengine binary
> > process shouldn't start connecting ICE candidates once the SIP part is
> > fully negotiated, which should trigger the rtpengine module on the
> > Kamailio to tell rtpengine binary.."ok..you can start associating now..."
>
> Not really. Why would you think that? ICE and SIP are really largely
> unrelated, and ICE processing starts as soon as possible so that media
> can flow as soon as possible. Any delay in ICE processing leads to
> delays in establishing the call or the media flows. (Trickle ICE exists
> partly for that reason.) In particular early media would break if ICE
> had to wait for SIP to finish.
>
> Cheers
>
>
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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

Hmm..that's interesting.  You would guess that the rtpengine binary process
shouldn't start connecting ICE candidates once the SIP part is fully
negotiated, which should trigger the rtpengine module on the Kamailio to
tell rtpengine binary.."ok..you can start associating now..."

On Fri, Dec 4, 2020 at 1:30 PM Richard Fuchs  wrote:

> On 04/12/2020 13.10, Andrew Chen wrote:
> > So from a SIP point of view, the 200 OK should of sent the final
> > negotiation of SDP once the client ACK's it right?
>
> The requirement to send an updated offer once ICE has completed with the
> final negotiated candidates existed in the original ICE RFC, but I
> believe it has been removed in the newer ICE RFC (I'm not 100% sure
> about that though). For rtpengine that requirement certainly doesn't
> exist and it's happy to negotiate and conclude ICE without a final
> updated offer. Also rtpengine itself cannot trigger such an updated
> offer since it's not directly in the SIP path, so that would have to be
> left up to the clients. So in short, a single offer/answer exchange is
> enough to get ICE started and it can complete negotiation out of band
> without further signalling.
>
> Cheers
>
>
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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

So from a SIP point of view, the 200 OK should of sent the final
negotiation of SDP once the client ACK's it right?

On Fri, Dec 4, 2020 at 11:49 AM Richard Fuchs  wrote:

> On 04/12/2020 11.39, Andrew Chen wrote:
> > oh...that's the IPv6 address of the STUN server, not the ipv6 of the
> > rtpengine instance.
>
> Ok. From rtpengine's point of view, this is one of the client's IP
> addresses. The ICE candidate in the SDP is the client telling rtpengine:
> This is one of my own addresses (`typ host`) and I can receive media
> here. It is also listed (by the client who has sent the SDP) as the
> highest priority candidate, and this is why it's (presumably) negotiated
> as the candidate to use. This is why rtpengine is sending media there.
>
> Cheers
>
>
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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

oh...that's the IPv6 address of the STUN server, not the ipv6 of the
rtpengine instance.

On Fri, Dec 4, 2020 at 11:38 AM Richard Fuchs  wrote:

> On 04/12/2020 11.31, Andrew Chen wrote:
> > If that's the case then I don't know why this line doesn't show the
> > ipv6 address of the client:
> >
> > Dec  3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG:
> > [ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint:
> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0)
>
> It doesn't?
>
> 2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]:
> a=candidate:3373280875 1 udp 2122262783
> 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0
> network-id 2 network-cost 10
>
> What is this then?
>
> Cheers
>
>
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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

If that's the case then I don't know why this line doesn't show the ipv6
address of the client:

Dec  3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG:
[ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint:
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0)

On Fri, Dec 4, 2020 at 11:25 AM Richard Fuchs  wrote:

> On 04/12/2020 10.39, Andrew Chen wrote:
> > So my next question would be:
> >
> > My 200 OK back to the client should have rtpengine as the only ICE
> > candidate.  Shouldn't it use that one instead?
>
> Yes it should. And it probably does.
>
> Cheers
>
>
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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

So my next question would be:

My 200 OK back to the client should have rtpengine as the only ICE
candidate.  Shouldn't it use that one instead?

On Fri, Dec 4, 2020 at 10:18 AM Richard Fuchs  wrote:

> On 04/12/2020 09.40, Andrew Chen wrote:
> > Hey Richard,
> >
> > So that's what I thought too until I saw this in rtpengine logs for
> > one of my test calls:
> >
> > Dec  3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG:
> > [ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint:
> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0)
> >
> > unless I misunderstood what this line means.
>
> I think you did. It means rtpengine has received a media packet from one
> side and has forwarded it to the other side. Of course it's one of the
> ICE candidates from the SDP because that's where the peer said it wants
> to receive the media from rtpengine.
>
> Cheers
>
>
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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Richard Fuchs]

On 04/12/2020 14.04, Andrew Chen wrote:
Sure...I understand ICE has its own setup workflow than SIP but it's 
also important that rtpengine uses the rtp path that's negotiated in 
the SIP or else it can cause confusion (to those who don't understand 
ICE very well like me).


There is no RTP path negotiated through SIP. There is only a number of 
possible paths, which are compiled from the list of candidates that each 
side advertises. ICE then checks which paths work, and then finally 
chooses the one with the highest priority.


The only way to have a single media path that is negotiated through SIP 
is if you don't use ICE.


Cheers



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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Richard Fuchs]

On 04/12/2020 13.36, Andrew Chen wrote:
Hmm..that's interesting.  You would guess that the rtpengine binary 
process shouldn't start connecting ICE candidates once the SIP part is 
fully negotiated, which should trigger the rtpengine module on the 
Kamailio to tell rtpengine binary.."ok..you can start associating now..."


Not really. Why would you think that? ICE and SIP are really largely 
unrelated, and ICE processing starts as soon as possible so that media 
can flow as soon as possible. Any delay in ICE processing leads to 
delays in establishing the call or the media flows. (Trickle ICE exists 
partly for that reason.) In particular early media would break if ICE 
had to wait for SIP to finish.


Cheers


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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

Hey Richard,

So that's what I thought too until I saw this in rtpengine logs for one of
my test calls:

Dec  3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG:
[ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint:
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0)

unless I misunderstood what this line means.

On Fri, Dec 4, 2020 at 9:34 AM Richard Fuchs  wrote:

> On 04/12/2020 09.24, Andrew Chen wrote:
> > Hey Richard,
> >
> > So it is true rtpengine is handling rtp between kamailio and receiver
> > (freeswitch).  I'm trying to understand if there is a way to not
> > forward rtp to any of the ICE candidates in the original INVITE
> > request from the client side.  In other words, have the client rtp
> > forward directly to the rtpengine and not any of the STUN servers (or
> > ICE candidates).  i honestly thought putting ICE=force in the
> > rtpengine_answer would help in this case...but I don't see this part
> > working.
>
> But that is what's happening, isn't it? What other media would rtpengine
> be sending to the client if not the media received from the other
> client? So your A client is sending to rtpengine, and rtpengine is
> forwarding that to B, while your B client is also sending to rtpengine,
> which then forwards it to A. Where do you not see that working?
>
> Cheers
>
>
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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Richard Fuchs]

On 04/12/2020 13.10, Andrew Chen wrote:
So from a SIP point of view, the 200 OK should of sent the final 
negotiation of SDP once the client ACK's it right?


The requirement to send an updated offer once ICE has completed with the 
final negotiated candidates existed in the original ICE RFC, but I 
believe it has been removed in the newer ICE RFC (I'm not 100% sure 
about that though). For rtpengine that requirement certainly doesn't 
exist and it's happy to negotiate and conclude ICE without a final 
updated offer. Also rtpengine itself cannot trigger such an updated 
offer since it's not directly in the SIP path, so that would have to be 
left up to the clients. So in short, a single offer/answer exchange is 
enough to get ICE started and it can complete negotiation out of band 
without further signalling.


Cheers


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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

Hey Richard,

So it is true rtpengine is handling rtp between kamailio and receiver
(freeswitch).  I'm trying to understand if there is a way to not
forward rtp to any of the ICE candidates in the original INVITE request
from the client side.  In other words, have the client rtp forward directly
to the rtpengine and not any of the STUN servers (or ICE candidates).  i
honestly thought putting ICE=force in the rtpengine_answer would help in
this case...but I don't see this part working.

On Fri, Dec 4, 2020 at 9:20 AM Richard Fuchs  wrote:

> On 04/12/2020 09.11, Andrew Chen wrote:
> > So for Yuriy's comment:
> >
> > I did issue ICE=force parameter but, as you can see my paste, it's
> > still sending RTP sequence packets to the ICE candidate, which is not
> > what I want to do.
> >
> > Richard,
> >
> > So our current setup is this:
> >
> > SIP
> > client -> kamailio -> freeswitch
> > RTP
> > client -> freeswitch
> >
> > What I want to accomplish is this:
> >
> > SIP
> > client -> kamailio -> freeswitch
> > RTP (bypass ICE)
> > client -> rtpengine -> freeswitch
>
> I still don't understand. What does "bypass ICE" mean?
>
> If you see rtpengine forwarding media packets as per the log you posted,
> what makes you think that media is not going through rtpengine?
>
> Cheers
>
>
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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

Actually Yuriy's..sorry...I see what you are saying.

So my question should be, this means I don't have a choice but to use the
ICE candidate between client and kamailio?

On Fri, Dec 4, 2020 at 9:11 AM Andrew Chen  wrote:

> So for Yuriy's comment:
>
> I did issue ICE=force parameter but, as you can see my paste, it's still
> sending RTP sequence packets to the ICE candidate, which is not what I want
> to do.
>
> Richard,
>
> So our current setup is this:
>
> SIP
> client -> kamailio -> freeswitch
> RTP
> client -> freeswitch
>
> What I want to accomplish is this:
>
> SIP
> client -> kamailio -> freeswitch
> RTP (bypass ICE)
> client -> rtpengine -> freeswitch
>
>
>
> On Fri, Dec 4, 2020 at 3:34 AM Yuriy Gorlichenko 
> wrote:
>
>> Hello
>> 2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]:
>> a=candidate:2274611867 1 tcp 1518283007
>> 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active generation
>> 0 network-id 2 network-cost 10
>> is a local candidate sent in the offer from the client
>> ICE=force just says rtpengine: "when you will resend this offer  to the
>> receiver, remove all candidates sent by sender and put yourself as the only
>> candidate"
>> But rtpengine still has to follow the ICE check for the received
>> candidates as in that case rtpengine is the receiver and it can't just not
>> check local candidates. That is why you facing this behaviour from the
>> rtpengine, which is totally correct.
>>
>> In other terms
>> Sender ---> rtpengine :rtpengine must follow ice procedure
>> and check availability of these local candidates to establish connection
>> between Sender and rtpengine
>> rtpengine -> Receiver:  rtpengine pretends to be a Sender for
>> the ICE agent of the receiver. but in this term, RTPengine will be the only
>> candidate for this part of the call, so this part is establishing a
>> connection between rtpengine and Receiver
>>
>> чт, 3 дек. 2020 г. в 23:40, Richard Fuchs :
>>
>>> On 03/12/2020 13.39, Andrew Chen wrote:
>>> > Hi all,
>>> >
>>> > I was wondering if someone can help me understand how the ICE
>>> > parameter works in the rtpengine module works.
>>> >
>>> > So basically our client does an ICE candidate lookup and grabs a list
>>> > of them and applies it to the INVITE that gets sent to the Kamailio.
>>> > The list looks like this:
>>> >
>>> > 2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]:
>>> > a=candidate:3373280875 1 udp 2122262783
>>> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0
>>> > network-id 2 network-cost 10
>>> > 2020-12-03T18:05:46.456057+00:00 ashmainkama51 kamailio[22147]:
>>> > a=candidate:3040609428 1 udp 2122197247
>>> > 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 59828 typ host generation 0
>>> > network-id 3 network-cost 10
>>> > 2020-12-03T18:05:46.456081+00:00 ashmainkama51 kamailio[22147]:
>>> > a=candidate:1681997092 1 udp 2122129151 192.168.1.66 59829 typ host
>>> > generation 0 network-id 1 network-cost 10
>>> > 2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]:
>>> > a=candidate:2274611867 1 tcp 1518283007
>>> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active
>>> > generation 0 network-id 2 network-cost 10
>>> > 2020-12-03T18:05:46.456131+00:00 ashmainkama51 kamailio[22147]:
>>> > a=candidate:4223662180 1 tcp 1518217471
>>> > 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 9 typ host tcptype active
>>> > generation 0 network-id 3 network-cost 10
>>> > 2020-12-03T18:05:46.456155+00:00 ashmainkama51 kamailio[22147]:
>>> > a=candidate:717406676 1 tcp 1518149375 192.168.1.66 9 typ host tcptype
>>> > active generation 0 network-id 1 network-cost 10
>>> > 2020-12-03T18:05:46.456180+00:00 ashmainkama51 kamailio[22147]:
>>> > a=candidate:2130547417 1 udp 8199935 206.81.191.27 61165 typ relay
>>> > raddr 85.247.0.121 rport 53091 generation 0 network-id 1 network-cost
>>> 10
>>> >
>>> >
>>> > In the rtpengine_offer, I have this:
>>> >
>>> > rtpengine_offer("ICE=force DTLS=passive replace-session-connection
>>> > replace-origin external internal");
>>> >
>>> > What i am trying to do is to tell the freeswitch endpoint to use
>>> > rtpengine as ICE candidate and I see in the SDP this is happening:
>>> >
>>> > a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host
>>> > generation 0
>>> >
>>> > Btw..206.81.191.52 is a separate AWS instance running rtpengine binary.
>>> >
>>> > On the rtpengine_answer, I tell the remote client side the same
>>> > thing..use rtpengine as your ICE candidate:
>>> >
>>> > rtpengine_answer("ICE=force DTLS=passive replace-session-connection
>>> > replace-origin internal external");
>>> >
>>> > and this is snippet from 200 OK:
>>> >
>>> > a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host
>>> > generation 0
>>> >
>>> > Question:
>>> >
>>> > Why does the rtpengine logs still show that it's trying to use
>>> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 as RTP candidate in this
>>> scenario?
>>> >
>>> > Dec  3 18:05:4

Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

So for Yuriy's comment:

I did issue ICE=force parameter but, as you can see my paste, it's still
sending RTP sequence packets to the ICE candidate, which is not what I want
to do.

Richard,

So our current setup is this:

SIP
client -> kamailio -> freeswitch
RTP
client -> freeswitch

What I want to accomplish is this:

SIP
client -> kamailio -> freeswitch
RTP (bypass ICE)
client -> rtpengine -> freeswitch



On Fri, Dec 4, 2020 at 3:34 AM Yuriy Gorlichenko 
wrote:

> Hello
> 2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]:
> a=candidate:2274611867 1 tcp 1518283007
> 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active generation
> 0 network-id 2 network-cost 10
> is a local candidate sent in the offer from the client
> ICE=force just says rtpengine: "when you will resend this offer  to the
> receiver, remove all candidates sent by sender and put yourself as the only
> candidate"
> But rtpengine still has to follow the ICE check for the received
> candidates as in that case rtpengine is the receiver and it can't just not
> check local candidates. That is why you facing this behaviour from the
> rtpengine, which is totally correct.
>
> In other terms
> Sender ---> rtpengine :rtpengine must follow ice procedure and
> check availability of these local candidates to establish connection
> between Sender and rtpengine
> rtpengine -> Receiver:  rtpengine pretends to be a Sender for
> the ICE agent of the receiver. but in this term, RTPengine will be the only
> candidate for this part of the call, so this part is establishing a
> connection between rtpengine and Receiver
>
> чт, 3 дек. 2020 г. в 23:40, Richard Fuchs :
>
>> On 03/12/2020 13.39, Andrew Chen wrote:
>> > Hi all,
>> >
>> > I was wondering if someone can help me understand how the ICE
>> > parameter works in the rtpengine module works.
>> >
>> > So basically our client does an ICE candidate lookup and grabs a list
>> > of them and applies it to the INVITE that gets sent to the Kamailio.
>> > The list looks like this:
>> >
>> > 2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]:
>> > a=candidate:3373280875 1 udp 2122262783
>> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0
>> > network-id 2 network-cost 10
>> > 2020-12-03T18:05:46.456057+00:00 ashmainkama51 kamailio[22147]:
>> > a=candidate:3040609428 1 udp 2122197247
>> > 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 59828 typ host generation 0
>> > network-id 3 network-cost 10
>> > 2020-12-03T18:05:46.456081+00:00 ashmainkama51 kamailio[22147]:
>> > a=candidate:1681997092 1 udp 2122129151 192.168.1.66 59829 typ host
>> > generation 0 network-id 1 network-cost 10
>> > 2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]:
>> > a=candidate:2274611867 1 tcp 1518283007
>> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active
>> > generation 0 network-id 2 network-cost 10
>> > 2020-12-03T18:05:46.456131+00:00 ashmainkama51 kamailio[22147]:
>> > a=candidate:4223662180 1 tcp 1518217471
>> > 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 9 typ host tcptype active
>> > generation 0 network-id 3 network-cost 10
>> > 2020-12-03T18:05:46.456155+00:00 ashmainkama51 kamailio[22147]:
>> > a=candidate:717406676 1 tcp 1518149375 192.168.1.66 9 typ host tcptype
>> > active generation 0 network-id 1 network-cost 10
>> > 2020-12-03T18:05:46.456180+00:00 ashmainkama51 kamailio[22147]:
>> > a=candidate:2130547417 1 udp 8199935 206.81.191.27 61165 typ relay
>> > raddr 85.247.0.121 rport 53091 generation 0 network-id 1 network-cost 10
>> >
>> >
>> > In the rtpengine_offer, I have this:
>> >
>> > rtpengine_offer("ICE=force DTLS=passive replace-session-connection
>> > replace-origin external internal");
>> >
>> > What i am trying to do is to tell the freeswitch endpoint to use
>> > rtpengine as ICE candidate and I see in the SDP this is happening:
>> >
>> > a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host
>> > generation 0
>> >
>> > Btw..206.81.191.52 is a separate AWS instance running rtpengine binary.
>> >
>> > On the rtpengine_answer, I tell the remote client side the same
>> > thing..use rtpengine as your ICE candidate:
>> >
>> > rtpengine_answer("ICE=force DTLS=passive replace-session-connection
>> > replace-origin internal external");
>> >
>> > and this is snippet from 200 OK:
>> >
>> > a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host
>> > generation 0
>> >
>> > Question:
>> >
>> > Why does the rtpengine logs still show that it's trying to use
>> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 as RTP candidate in this scenario?
>> >
>> > Dec  3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG:
>> > [ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint:
>> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0)
>> >
>> > I thought ICE=force will handle such adjustments so that all rtp is
>> > handled by the rtpengine to our client and bypass the STUN server?
>> >
>> Isn't that what's happening? Rtpengine receiv

Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Richard Fuchs]

On 04/12/2020 11.39, Andrew Chen wrote:
oh...that's the IPv6 address of the STUN server, not the ipv6 of the 
rtpengine instance.


Ok. From rtpengine's point of view, this is one of the client's IP 
addresses. The ICE candidate in the SDP is the client telling rtpengine: 
This is one of my own addresses (`typ host`) and I can receive media 
here. It is also listed (by the client who has sent the SDP) as the 
highest priority candidate, and this is why it's (presumably) negotiated 
as the candidate to use. This is why rtpengine is sending media there.


Cheers


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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Richard Fuchs]

On 04/12/2020 11.31, Andrew Chen wrote:
If that's the case then I don't know why this line doesn't show the 
ipv6 address of the client:


Dec  3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG: 
[ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint: 
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0)


It doesn't?

2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]: 
a=candidate:3373280875 1 udp 2122262783 
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0 
network-id 2 network-cost 10


What is this then?

Cheers


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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Richard Fuchs]

On 04/12/2020 10.39, Andrew Chen wrote:

So my next question would be:

My 200 OK back to the client should have rtpengine as the only ICE 
candidate.  Shouldn't it use that one instead?


Yes it should. And it probably does.

Cheers


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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Richard Fuchs]

On 04/12/2020 09.40, Andrew Chen wrote:

Hey Richard,

So that's what I thought too until I saw this in rtpengine logs for 
one of my test calls:


Dec  3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG: 
[ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint: 
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0)


unless I misunderstood what this line means.


I think you did. It means rtpengine has received a media packet from one 
side and has forwarded it to the other side. Of course it's one of the 
ICE candidates from the SDP because that's where the peer said it wants 
to receive the media from rtpengine.


Cheers


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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Richard Fuchs]

On 04/12/2020 09.24, Andrew Chen wrote:

Hey Richard,

So it is true rtpengine is handling rtp between kamailio and receiver 
(freeswitch).  I'm trying to understand if there is a way to not 
forward rtp to any of the ICE candidates in the original INVITE 
request from the client side.  In other words, have the client rtp 
forward directly to the rtpengine and not any of the STUN servers (or 
ICE candidates).  i honestly thought putting ICE=force in the 
rtpengine_answer would help in this case...but I don't see this part 
working.


But that is what's happening, isn't it? What other media would rtpengine 
be sending to the client if not the media received from the other 
client? So your A client is sending to rtpengine, and rtpengine is 
forwarding that to B, while your B client is also sending to rtpengine, 
which then forwards it to A. Where do you not see that working?


Cheers


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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Richard Fuchs]

On 04/12/2020 09.11, Andrew Chen wrote:

So for Yuriy's comment:

I did issue ICE=force parameter but, as you can see my paste, it's 
still sending RTP sequence packets to the ICE candidate, which is not 
what I want to do.


Richard,

So our current setup is this:

SIP
client -> kamailio -> freeswitch
RTP
client -> freeswitch

What I want to accomplish is this:

SIP
client -> kamailio -> freeswitch
RTP (bypass ICE)
client -> rtpengine -> freeswitch


I still don't understand. What does "bypass ICE" mean?

If you see rtpengine forwarding media packets as per the log you posted, 
what makes you think that media is not going through rtpengine?


Cheers


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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Yuriy Gorlichenko]

Hello
2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]:
a=candidate:2274611867 1 tcp 1518283007
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active generation
0 network-id 2 network-cost 10
is a local candidate sent in the offer from the client
ICE=force just says rtpengine: "when you will resend this offer  to the
receiver, remove all candidates sent by sender and put yourself as the only
candidate"
But rtpengine still has to follow the ICE check for the received candidates
as in that case rtpengine is the receiver and it can't just not check local
candidates. That is why you facing this behaviour from the rtpengine, which
is totally correct.

In other terms
Sender ---> rtpengine :rtpengine must follow ice procedure and
check availability of these local candidates to establish connection
between Sender and rtpengine
rtpengine -> Receiver:  rtpengine pretends to be a Sender for
the ICE agent of the receiver. but in this term, RTPengine will be the only
candidate for this part of the call, so this part is establishing a
connection between rtpengine and Receiver

чт, 3 дек. 2020 г. в 23:40, Richard Fuchs :

> On 03/12/2020 13.39, Andrew Chen wrote:
> > Hi all,
> >
> > I was wondering if someone can help me understand how the ICE
> > parameter works in the rtpengine module works.
> >
> > So basically our client does an ICE candidate lookup and grabs a list
> > of them and applies it to the INVITE that gets sent to the Kamailio.
> > The list looks like this:
> >
> > 2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]:
> > a=candidate:3373280875 1 udp 2122262783
> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0
> > network-id 2 network-cost 10
> > 2020-12-03T18:05:46.456057+00:00 ashmainkama51 kamailio[22147]:
> > a=candidate:3040609428 1 udp 2122197247
> > 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 59828 typ host generation 0
> > network-id 3 network-cost 10
> > 2020-12-03T18:05:46.456081+00:00 ashmainkama51 kamailio[22147]:
> > a=candidate:1681997092 1 udp 2122129151 192.168.1.66 59829 typ host
> > generation 0 network-id 1 network-cost 10
> > 2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]:
> > a=candidate:2274611867 1 tcp 1518283007
> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active
> > generation 0 network-id 2 network-cost 10
> > 2020-12-03T18:05:46.456131+00:00 ashmainkama51 kamailio[22147]:
> > a=candidate:4223662180 1 tcp 1518217471
> > 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 9 typ host tcptype active
> > generation 0 network-id 3 network-cost 10
> > 2020-12-03T18:05:46.456155+00:00 ashmainkama51 kamailio[22147]:
> > a=candidate:717406676 1 tcp 1518149375 192.168.1.66 9 typ host tcptype
> > active generation 0 network-id 1 network-cost 10
> > 2020-12-03T18:05:46.456180+00:00 ashmainkama51 kamailio[22147]:
> > a=candidate:2130547417 1 udp 8199935 206.81.191.27 61165 typ relay
> > raddr 85.247.0.121 rport 53091 generation 0 network-id 1 network-cost 10
> >
> >
> > In the rtpengine_offer, I have this:
> >
> > rtpengine_offer("ICE=force DTLS=passive replace-session-connection
> > replace-origin external internal");
> >
> > What i am trying to do is to tell the freeswitch endpoint to use
> > rtpengine as ICE candidate and I see in the SDP this is happening:
> >
> > a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host
> > generation 0
> >
> > Btw..206.81.191.52 is a separate AWS instance running rtpengine binary.
> >
> > On the rtpengine_answer, I tell the remote client side the same
> > thing..use rtpengine as your ICE candidate:
> >
> > rtpengine_answer("ICE=force DTLS=passive replace-session-connection
> > replace-origin internal external");
> >
> > and this is snippet from 200 OK:
> >
> > a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host
> > generation 0
> >
> > Question:
> >
> > Why does the rtpengine logs still show that it's trying to use
> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 as RTP candidate in this scenario?
> >
> > Dec  3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG:
> > [ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint:
> > 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0)
> >
> > I thought ICE=force will handle such adjustments so that all rtp is
> > handled by the rtpengine to our client and bypass the STUN server?
> >
> Isn't that what's happening? Rtpengine receives media from one peer and
> forwards it to one of the ICE candidates (presumably the primary one
> that was negotiated) of the other peer. Or am I misunderstanding what
> you posted?
>
> Cheers
>
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[SR-Users] question regarding rtpengine and ICE candidate selection

[Andrew Chen]

Hi all,

I was wondering if someone can help me understand how the ICE parameter
works in the rtpengine module works.

So basically our client does an ICE candidate lookup and grabs a list of
them and applies it to the INVITE that gets sent to the Kamailio.  The list
looks like this:

2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]:
a=candidate:3373280875 1 udp 2122262783
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0
network-id 2 network-cost 10
2020-12-03T18:05:46.456057+00:00 ashmainkama51 kamailio[22147]:
a=candidate:3040609428 1 udp 2122197247
2001:8a0:78fc:7000:d979:bf75:dbc0:69f 59828 typ host generation 0
network-id 3 network-cost 10
2020-12-03T18:05:46.456081+00:00 ashmainkama51 kamailio[22147]:
a=candidate:1681997092 1 udp 2122129151 192.168.1.66 59829 typ host
generation 0 network-id 1 network-cost 10
2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]:
a=candidate:2274611867 1 tcp 1518283007
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active generation
0 network-id 2 network-cost 10
2020-12-03T18:05:46.456131+00:00 ashmainkama51 kamailio[22147]:
a=candidate:4223662180 1 tcp 1518217471
2001:8a0:78fc:7000:d979:bf75:dbc0:69f 9 typ host tcptype active generation
0 network-id 3 network-cost 10
2020-12-03T18:05:46.456155+00:00 ashmainkama51 kamailio[22147]:
a=candidate:717406676 1 tcp 1518149375 192.168.1.66 9 typ host tcptype
active generation 0 network-id 1 network-cost 10
2020-12-03T18:05:46.456180+00:00 ashmainkama51 kamailio[22147]:
a=candidate:2130547417 1 udp 8199935 206.81.191.27 61165 typ relay raddr
85.247.0.121 rport 53091 generation 0 network-id 1 network-cost 10


In the rtpengine_offer, I have this:

rtpengine_offer("ICE=force DTLS=passive replace-session-connection
replace-origin external internal");

What i am trying to do is to tell the freeswitch endpoint to use rtpengine
as ICE candidate and I see in the SDP this is happening:

a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host generation
0

Btw..206.81.191.52 is a separate AWS instance running rtpengine binary.

On the rtpengine_answer, I tell the remote client side the same thing..use
rtpengine as your ICE candidate:

rtpengine_answer("ICE=force DTLS=passive replace-session-connection
replace-origin internal external");

and this is snippet from 200 OK:

a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host generation
0

Question:

Why does the rtpengine logs still show that it's trying to use
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 as RTP candidate in this scenario?

Dec  3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG:
[ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint:
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0)

I thought ICE=force will handle such adjustments so that all rtp is handled
by the rtpengine to our client and bypass the STUN server?



Thanks.

-- 
Andy Chen
Sr. Telephony Lead Engineer
achen@ fuze.com

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Re: [SR-Users] question regarding rtpengine and ICE candidate selection

[Richard Fuchs]

On 03/12/2020 13.39, Andrew Chen wrote:

Hi all,

I was wondering if someone can help me understand how the ICE 
parameter works in the rtpengine module works.


So basically our client does an ICE candidate lookup and grabs a list 
of them and applies it to the INVITE that gets sent to the Kamailio.  
The list looks like this:


2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]: 
a=candidate:3373280875 1 udp 2122262783 
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0 
network-id 2 network-cost 10
2020-12-03T18:05:46.456057+00:00 ashmainkama51 kamailio[22147]: 
a=candidate:3040609428 1 udp 2122197247 
2001:8a0:78fc:7000:d979:bf75:dbc0:69f 59828 typ host generation 0 
network-id 3 network-cost 10
2020-12-03T18:05:46.456081+00:00 ashmainkama51 kamailio[22147]: 
a=candidate:1681997092 1 udp 2122129151 192.168.1.66 59829 typ host 
generation 0 network-id 1 network-cost 10
2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]: 
a=candidate:2274611867 1 tcp 1518283007 
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active 
generation 0 network-id 2 network-cost 10
2020-12-03T18:05:46.456131+00:00 ashmainkama51 kamailio[22147]: 
a=candidate:4223662180 1 tcp 1518217471 
2001:8a0:78fc:7000:d979:bf75:dbc0:69f 9 typ host tcptype active 
generation 0 network-id 3 network-cost 10
2020-12-03T18:05:46.456155+00:00 ashmainkama51 kamailio[22147]: 
a=candidate:717406676 1 tcp 1518149375 192.168.1.66 9 typ host tcptype 
active generation 0 network-id 1 network-cost 10
2020-12-03T18:05:46.456180+00:00 ashmainkama51 kamailio[22147]: 
a=candidate:2130547417 1 udp 8199935 206.81.191.27 61165 typ relay 
raddr 85.247.0.121 rport 53091 generation 0 network-id 1 network-cost 10



In the rtpengine_offer, I have this:

rtpengine_offer("ICE=force DTLS=passive replace-session-connection 
replace-origin external internal");


What i am trying to do is to tell the freeswitch endpoint to use 
rtpengine as ICE candidate and I see in the SDP this is happening:


a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host 
generation 0


Btw..206.81.191.52 is a separate AWS instance running rtpengine binary.

On the rtpengine_answer, I tell the remote client side the same 
thing..use rtpengine as your ICE candidate:


rtpengine_answer("ICE=force DTLS=passive replace-session-connection 
replace-origin internal external");


and this is snippet from 200 OK:

a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host 
generation 0


Question:

Why does the rtpengine logs still show that it's trying to use 
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 as RTP candidate in this scenario?


Dec  3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG: 
[ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint: 
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0)


I thought ICE=force will handle such adjustments so that all rtp is 
handled by the rtpengine to our client and bypass the STUN server?


Isn't that what's happening? Rtpengine receives media from one peer and 
forwards it to one of the ICE candidates (presumably the primary one 
that was negotiated) of the other peer. Or am I misunderstanding what 
you posted?


Cheers

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Re: [SR-Users] Question about textops

[Henning Westerholt]

Hi Mojtaba,

not looked in the code – but it could be related to be able to check with this 
function for request and replies. Replies will have the method name in their 
CSEQ.

If you have a requirement that is not available right now in the module, it can 
be of course extended (e.g. by pull request).

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>

From: sr-users  On Behalf Of Mojtaba
Sent: Friday, June 26, 2020 12:43 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Question about textops

Any idea?
On Wed, 24 Jun 2020, 14:26 Mojtaba, mailto:mes...@gmail.com>> 
wrote:
Hello,
Some confusing things while developing in Kamailio:
In is_method_f function in textops module, the code checks the HDR_CSEQ_F in 
msg also, and if the method name appears in CSEQ_F, The result is returned TRUE.
These question will  arise why in this function, the CSEQ field in message will 
check?
if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL)

{

LM_ERR("cannot parse cseq header\n");

return -1; /* should it be 0 ?!?! */

}

if(m->s==0)

return (get_cseq(msg)->method_id&m->len)?1:-1;

else

return (get_cseq(msg)->method_id==METHOD_OTHER

&& get_cseq(msg)->method.len==m->len

&& (strncasecmp(get_cseq(msg)->method.s, m->s,
m->len)==0))?1:-1;

I think it would be nice to add new function like: is_cseq_method_f in module 
to avoid some problem understanding.

--
--Mojtaba Esfandiari.S
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Re: [SR-Users] Question about textops

[Mojtaba]

Any idea?

On Wed, 24 Jun 2020, 14:26 Mojtaba,  wrote:

> Hello,
> Some confusing things while developing in Kamailio:
> In is_method_f function in textops module, the code checks the HDR_CSEQ_F
> in msg also, and if the method name appears in CSEQ_F, The result is
> returned TRUE.
> These question will  arise why in this function, the CSEQ field in message
> will check?
> if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL)
> {
> LM_ERR("cannot parse cseq header\n");
> return -1; /* should it be 0 ?!?! */
> }
> if(m->s==0)
> return (get_cseq(msg)->method_id&m->len)?1:-1;
> else
> return (get_cseq(msg)->method_id==METHOD_OTHER
> && get_cseq(msg)->method.len==m->len
> && (strncasecmp(get_cseq(msg)->method.s, m->s,
> m->len)==0))?1:-1;
>
> I think it would be nice to add new function like: is_cseq_method_f in
> module to avoid some problem understanding.
>
> --
> --Mojtaba Esfandiari.S
>
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[SR-Users] Question about textops

[Mojtaba]

Hello,
Some confusing things while developing in Kamailio:
In is_method_f function in textops module, the code checks the HDR_CSEQ_F
in msg also, and if the method name appears in CSEQ_F, The result is
returned TRUE.
These question will  arise why in this function, the CSEQ field in message
will check?
if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL)
{
LM_ERR("cannot parse cseq header\n");
return -1; /* should it be 0 ?!?! */
}
if(m->s==0)
return (get_cseq(msg)->method_id&m->len)?1:-1;
else
return (get_cseq(msg)->method_id==METHOD_OTHER
&& get_cseq(msg)->method.len==m->len
&& (strncasecmp(get_cseq(msg)->method.s, m->s,
m->len)==0))?1:-1;

I think it would be nice to add new function like: is_cseq_method_f in
module to avoid some problem understanding.

-- 
--Mojtaba Esfandiari.S
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Re: [SR-Users] Question about xhttp module kamailio 5.2.x

[Henning Westerholt]

Hello Jimmy,

not being the author of the module, so some generic comments.

The xhttp module is used from the jsonrpcs module, but the jsonrpcs module 
controls the handling. This module implements the asynchronous processing as 
you quoted below. The xhttp module does not implement asynchronous handling, so 
it will block if you use it stand alone.

I would suggest to just test if your worker process block in your particular 
situation and configuration. You can e.g. use the sleep() function of cfgutils 
to simulate a long running execution. See if you still are able to process SIP 
messages after you trigger n requests (for n = kamailio children count).

Cheers,

Henning

Am 16.01.20 um 23:02 schrieb Jimmy Rodley:
Hi,

I am on kamailio 5.2.4 and want to use the xhttp module for certain management 
operations.

The request involves some DB queries . The DB is remote and via ODBC and might 
have some delay.
The documentation on the xhttp module states about latency 
http://kamailio.org/docs/modules/devel/modules/xhttp.html
 mentioning that the request is handled in an asynchronous manner.

However, the JSONRPC-S module documentation says that this module implements 
the support for asynchronous RPC commands only for HTTP and HTTPS transports.

https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm1049416972

I am confused about the process under whose context the xttp request . Can 
there be a dedicated process to handle certain management requests which do 
some custom instructions like certain DB operations or initiate certain http 
requests ?

Thanks,
Rupesh





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--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://gilawa.com
Kamailio Merchandising - https://skalatan.de/merchandising/
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[SR-Users] Question about xhttp module kamailio 5.2.x

[Jimmy Rodley]

Hi,
I am on kamailio 5.2.4 and want to use the xhttp module for certain management 
operations.
The request involves some DB queries . The DB is remote and via ODBC and might 
have some delay.The documentation on the xhttp module states about latency 
http://kamailio.org/docs/modules/devel/modules/xhttp.html mentioning that the 
request is handled in an asynchronous manner. 
However, the JSONRPC-S module documentation says that this module implements 
the support for asynchronous RPC commands only for HTTP and HTTPS transports.

https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm1049416972

I am confused about the process under whose context the xttp request . Can 
there be a dedicated process to handle certain management requests which do 
some custom instructions like certain DB operations or initiate certain http 
requests ?
Thanks,Rupesh

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Re: [SR-Users] Question regarding priority in dispatcher module

[Floimair Florian]

Hi Henning!

Yes I was referring to algorithm 8.

Thanks for the quick clarification!


With best regards

FLORIAN FLOIMAIR

Software Development
Commend International GmbH
Saalachstrasse 51
5020 Salzburg, Austria
commend.com
LG Salzburg / FN 178618z
 
 

Am 05.12.19, 15:35 schrieb "Henning Westerholt" :

Hi Florian,

If you are referring to algorithm 8 in dispatcher - highest priority 
numerical value is chosen first.

Cheers,

Henning

-- 
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https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fskalatan.de%2Fblog%2F&data=02%7C01%7Cf.floimair%40commend.com%7Cb26fa49c351b49ce55e408d7799068cc%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637111533448085251&sdata=%2BGzONxt8MqzY%2BwhVpoaBI9KwFWiDCsTfhxNV1WQVLjk%3D&reserved=0
Kamailio services – 
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-Original Message-
From: sr-users  On Behalf Of Floimair 
Florian
Sent: Thursday, December 5, 2019 3:22 PM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Question regarding priority in dispatcher module

Dear all!

After reading the documentation on dispatcher module I still have a 
question that isn't really documented.
Regarding the priority value of an entry in a dispatcher list:

Is a lower numerical priority value regarded as higher priority or is it 
the other way around?

Hopefully someone can help me to clarify this.

With best regards

FLORIAN FLOIMAIR

Software Development
Commend International GmbH
Saalachstrasse 51
5020 Salzburg, Austria
commend.com
LG Salzburg / FN 178618z
 
 

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Re: [SR-Users] Question regarding priority in dispatcher module

[Henning Westerholt]

Hi Florian,

If you are referring to algorithm 8 in dispatcher - highest priority numerical 
value is chosen first.

Cheers,

Henning

-- 
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com 

-Original Message-
From: sr-users  On Behalf Of Floimair 
Florian
Sent: Thursday, December 5, 2019 3:22 PM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Question regarding priority in dispatcher module

Dear all!

After reading the documentation on dispatcher module I still have a question 
that isn't really documented.
Regarding the priority value of an entry in a dispatcher list:

Is a lower numerical priority value regarded as higher priority or is it the 
other way around?

Hopefully someone can help me to clarify this.

With best regards

FLORIAN FLOIMAIR

Software Development
Commend International GmbH
Saalachstrasse 51
5020 Salzburg, Austria
commend.com
LG Salzburg / FN 178618z
 
 

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[SR-Users] Question regarding priority in dispatcher module

[Floimair Florian]

Dear all!

After reading the documentation on dispatcher module I still have a question 
that isn't really documented.
Regarding the priority value of an entry in a dispatcher list:

Is a lower numerical priority value regarded as higher priority or is it the 
other way around?

Hopefully someone can help me to clarify this.

With best regards

FLORIAN FLOIMAIR

Software Development
Commend International GmbH
Saalachstrasse 51
5020 Salzburg, Austria
commend.com
LG Salzburg / FN 178618z
 
 

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[SR-Users] Question about ims_ipsec_pcscf

[Woscek, Martin W.]

Hi all,

We are looking for information as to what the ims_ipsec_pcscf module was 
written for and tested against.

What was the target UE/EPC/PCRF that Tsvetomir Dimitrov wrote the module 
towards?
Has an emulated EPC been tested with this module?  If so which one(s)?

Thanks in advance,
Martin

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Re: [SR-Users] Question on Database Version Table

[David Villasmil]

Nevermind, got it.

~# grep "INSERT.*version" /usr/share/kamailio/mysql/*
> /usr/share/kamailio/mysql/acc-create.sql:INSERT INTO version (table_name,
> table_version) values ('acc','5');
> /usr/share/kamailio/mysql/acc-create.sql:INSERT INTO version (table_name,
> table_version) values ('acc_cdrs','2');
> /usr/share/kamailio/mysql/acc-create.sql:INSERT INTO version (table_name,
> table_version) values ('missed_calls','4');
> /usr/share/kamailio/mysql/alias_db-create.sql:INSERT INTO version
> (table_name, table_version) values ('dbaliases','1');
> /usr/share/kamailio/mysql/auth_db-create.sql:INSERT INTO version
> (table_name, table_version) values ('subscriber','7');
> /usr/share/kamailio/mysql/avpops-create.sql:INSERT INTO version
> (table_name, table_version) values ('usr_preferences','2');
> /usr/share/kamailio/mysql/carrierroute-create.sql:INSERT INTO version
> (table_name, table_version) values ('carrierroute','3');
> /usr/share/kamailio/mysql/carrierroute-create.sql:INSERT INTO version
> (table_name, table_version) values ('carrierfailureroute','2');
> /usr/share/kamailio/mysql/carrierroute-create.sql:INSERT INTO version
> (table_name, table_version) values ('carrier_name','1');
> /usr/share/kamailio/mysql/carrierroute-create.sql:INSERT INTO version
> (table_name, table_version) values ('domain_name','1');
> /usr/share/kamailio/mysql/cpl-create.sql:INSERT INTO version (table_name,
> table_version) values ('cpl','1');
> /usr/share/kamailio/mysql/dialog-create.sql:INSERT INTO version
> (table_name, table_version) values ('dialog','7');
> /usr/share/kamailio/mysql/dialog-create.sql:INSERT INTO version
> (table_name, table_version) values ('dialog_vars','1');
> /usr/share/kamailio/mysql/dialplan-create.sql:INSERT INTO version
> (table_name, table_version) values ('dialplan','2');
> /usr/share/kamailio/mysql/dispatcher-create.sql:INSERT INTO version
> (table_name, table_version) values ('dispatcher','4');
> /usr/share/kamailio/mysql/domain-create.sql:INSERT INTO version
> (table_name, table_version) values ('domain','2');
> /usr/share/kamailio/mysql/domain-create.sql:INSERT INTO version
> (table_name, table_version) values ('domain_attrs','1');
> /usr/share/kamailio/mysql/domainpolicy-create.sql:INSERT INTO version
> (table_name, table_version) values ('domainpolicy','2');
> /usr/share/kamailio/mysql/drouting-create.sql:INSERT INTO version
> (table_name, table_version) values ('dr_gateways','3');
> /usr/share/kamailio/mysql/drouting-create.sql:INSERT INTO version
> (table_name, table_version) values ('dr_rules','3');
> /usr/share/kamailio/mysql/drouting-create.sql:INSERT INTO version
> (table_name, table_version) values ('dr_gw_lists','1');
> /usr/share/kamailio/mysql/drouting-create.sql:INSERT INTO version
> (table_name, table_version) values ('dr_groups','2');
> /usr/share/kamailio/mysql/group-create.sql:INSERT INTO version
> (table_name, table_version) values ('grp','2');
> /usr/share/kamailio/mysql/group-create.sql:INSERT INTO version
> (table_name, table_version) values ('re_grp','1');
> /usr/share/kamailio/mysql/htable-create.sql:INSERT INTO version
> (table_name, table_version) values ('htable','2');
> /usr/share/kamailio/mysql/imc-create.sql:INSERT INTO version (table_name,
> table_version) values ('imc_rooms','1');
> /usr/share/kamailio/mysql/imc-create.sql:INSERT INTO version (table_name,
> table_version) values ('imc_members','1');
> /usr/share/kamailio/mysql/ims_charging_create.sql:INSERT INTO version
> (table_name, table_version) values ('ro_session','6');
> /usr/share/kamailio/mysql/ims_dialog-create.sql:INSERT INTO version
> (table_name, table_version) values ('dialog_in', 7), ('dialog_out', 7),
> ('dialog_vars', 7);
> /usr/share/kamailio/mysql/ims_usrloc_pcscf-create.sql:INSERT INTO version
> (table_name, table_version) values ('location','6');
> /usr/share/kamailio/mysql/ims_usrloc_scscf-create.sql:INSERT INTO version
> (table_name, table_version) values ('contact','6');
> /usr/share/kamailio/mysql/ims_usrloc_scscf-create.sql:INSERT INTO version
> (table_name, table_version) values ('impu','6');
> /usr/share/kamailio/mysql/ims_usrloc_scscf-create.sql:INSERT INTO version
> (table_name, table_version) values ('impu_contact','6');
> /usr/share/kamailio/mysql/ims_usrloc_scscf-create.sql:INSERT INTO version
> (table_name, table_version) values ('subscriber','6');
> /usr/share/kamailio/mysql/ims_usrloc_scscf-create.sql:INSERT INTO version
> (table_name, table_version) values ('impu_subscriber','6');
> /usr/share/kamailio/mysql/lcr-create.sql:INSERT INTO version (table_name,
> table_version) values ('lcr_gw','3');
> /usr/share/kamailio/mysql/lcr-create.sql:INSERT INTO version (table_name,
> table_version) values ('lcr_rule_target','1');
> /usr/share/kamailio/mysql/lcr-create.sql:INSERT INTO version (table_name,
> table_version) values ('lcr_rule','3');
> /usr/share/kamailio/mysql/matrix-create.sql:INSERT INTO version
> (table_name, table_version) values ('matrix','1');
> /usr/share/kamailio/mysql

Re: [SR-Users] Question on Database Version Table

[David Villasmil]

Hello All,

I need to do a version upgrade from 5 to 5.2. I read about the
version_table parameter and i know i can create it and set the versions
there.
But where do i get the correct versions for the table? i.e.: location is
version N?

Thanks all!


Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Wed, Jun 19, 2019 at 8:35 AM Daniel-Constantin Mierla 
wrote:

>
> On 18.06.19 22:11, Alex Balashov wrote:
> > On Tue, Jun 18, 2019 at 10:09:56PM +0200, Daniel-Constantin Mierla wrote:
> >
> >> Where did you meet such situation?
> > I wish I could remember. It's been in various times over the years, but
> > I never filed a bug report--probably because I didn't really perceive it
> > as a bug consciously. :-) If I had to guess, I think it was in usrloc.
>
> I do not remember such issue reported or encountered for usrloc module,
> which is one quite popular. But it gets close to 20 years for usrloc, so
> a lot could have happened along the time ...
>
> The version table is rather ancient basic check just to try to avoid
> having kamailio starting ok, and then at runtime printing a lot of error
> messages and not functioning properly. Probably it should be improved
> somehow -- db api v2 tried to come with another approach, but that
> direction was not followed much.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla -- www.asipto.com
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>
>
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Re: [SR-Users] Question

[Alex Balashov]

Oh yeah. 

—
Sent from mobile, with due apologies for brevity and errors.

> On Sep 24, 2019, at 7:32 PM, Shahdee Nelson  wrote:
> 
> With your software would I be able to create, and modify VOIP lines? As in 
> saying would I be able to provide VOIP numbers that can make outbound calls 
> to USA from USA?
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[SR-Users] Question

[Shahdee Nelson]

With your software would I be able to create, and modify VOIP lines? As in
saying would I be able to provide VOIP numbers that can make outbound calls
to USA from USA?
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Re: [SR-Users] Question regarding Dispatcher + ds_default_socket() modparam + Call-ID generation

[Henning Westerholt]

Hi Daniel,

good tip about the crypto module.

It will indeed hide the IP address in the call-id, just tried it:

Call-ID: d810d369-aa60-4d79-aee5-9c03d2e66781

I think if we would feed the sruid unique ID (in libs/srutils) into the SHA256 
hash (also available there) this it should be secure enough without the crypto 
dependencies.

The sruid is based on a LSFR implementation which is quite fast, and the SHA256 
should provide the necessary unpredictability.

But as the crypto module approach works, it has probably not a really high 
priority. I can also look a bit into it when I got time.

Cheers,

Henning

Am 14.08.19 um 19:21 schrieb Daniel-Constantin Mierla:

Hello,

using the ip in call-id is not a good practice imo, I had it in mind to replace 
it properly everywhere for quite some time -- actually at this moment there is 
an option that can be activated in the crypto module making the call-id to be 
generated with libssl unique id generation functions, but I don't recall if the 
local ip is still appended. This would require libssl, so my goal was to add an 
alternative to generate a "good-enough" unique id, without external 
dependencies, to be used as local call-id.

Cheers,
Daniel

On 14.08.19 19:01, Joel Serrano wrote:
Hello Henning,

No concerns at all!! As you say, the Call-ID can really say whatever...  The 
only concern could/would be in the security topic that you are disclosing 
potential sensible information about your infrastructure blablabla... but that 
can be solved just by changing the listen= order so even that wouldn't be a 
problem.. In reality I was just curious so I thought I'd ask :)

Thanks!!
Joel.



On Wed, Aug 14, 2019 at 9:32 AM Henning Westerholt 
mailto:h...@skalatan.de>> wrote:

Hello Joel,

funny - I just had this discussion about the same topic some days ago.

In the end this is "only" the call-id, the IP should not be used to to routing 
descisions etc.. Do you have some more concerns about this?

I think as well it just uses the first IP. I think at the moment the call-id is 
generated internally from tm, but this could be of course changed for the 
module with some coding/extension.

Cheers,

Henning

Am 14.08.19 um 17:12 schrieb Joel Serrano:
Hello,

Simple doubt regarding dispatcher module when you have the ds_default_socket() 
modparam set:

Say you have 2 kamailio boxes, active + passive, one shared IP via 
keepalived On the actrive node, you have dispatcher.send_ping to 1, and the 
passive has it set to 0.

So far OK, only the active box is sending out probing OPTIONS requests, and 
it's using as outbound socket the virtual IP out of all the available ones (as 
defined with ds_default_socket() modparam)

I have seen in a capture that the outbound OPTIONS look like:

OPTIONS sip:190.14.203.219:5060 SIP/2.0
Via: SIP/2.0/UDP A.B.C.180;branch=z9hG4bK7abb.6fbaccc1.0
To: http://190.14.203.219:5060>>
From: 
;tag=e2bdd495733c59fd91487a137fccad4e-8a73
CSeq: 10 OPTIONS
Call-ID: 2019f8491329c382-31419@A.B.C.181
Max-Forwards: 70
Content-Length: 0

A.B.C.180 -> VRRP
A.B.C.181 -> Physical IP of the box

My doubt is, shouldn't ds_default_socket also update the IP used to generate 
the Call-ID used in the OPTIONS request? Currently, I believe it's using the 
first defined listen= IP from config script as I switched the order the listen= 
are defined and I get the desired result:

OPTIONS sip:186.188.220.174:5060 SIP/2.0
Via: SIP/2.0/UDP A.B.C.180;branch=z9hG4bKc97e.a8d9e1c2.0
To: http://186.188.220.174:5060>>
From: 
;tag=5e2e1773f812f6a7e4085c5d036e29d8-d323
CSeq: 10 OPTIONS
Call-ID: 7d9a92c218fc1ba0-32111@A.B.C.180
Max-Forwards: 70
Content-Length: 0


Is this expected?

Cheers,
Joel.






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www.twitter.com/miconda -- 
www.linkedin.com/in/miconda

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Re: [SR-Users] Question regarding Dispatcher + ds_default_socket() modparam + Call-ID generation

[Daniel-Constantin Mierla]

Hello,

using the ip in call-id is not a good practice imo, I had it in mind to
replace it properly everywhere for quite some time -- actually at this
moment there is an option that can be activated in the crypto module
making the call-id to be generated with libssl unique id generation
functions, but I don't recall if the local ip is still appended. This
would require libssl, so my goal was to add an alternative to generate a
"good-enough" unique id, without external dependencies, to be used as
local call-id.

Cheers,
Daniel

On 14.08.19 19:01, Joel Serrano wrote:
> Hello Henning, 
>
> No concerns at all!! As you say, the Call-ID can really say
> whatever...  The only concern could/would be in the security topic
> that you are disclosing potential sensible information about your
> infrastructure blablabla... but that can be solved just by changing
> the listen= order so even that wouldn't be a problem.. In reality I
> was just curious so I thought I'd ask :)
>
> Thanks!!
> Joel.
>
>
>
> On Wed, Aug 14, 2019 at 9:32 AM Henning Westerholt  > wrote:
>
> Hello Joel,
>
> funny - I just had this discussion about the same topic some days ago.
>
> In the end this is "only" the call-id, the IP should not be used
> to to routing descisions etc.. Do you have some more concerns
> about this?
>
> I think as well it just uses the first IP. I think at the moment
> the call-id is generated internally from tm, but this could be of
> course changed for the module with some coding/extension.
>
> Cheers,
>
> Henning
>
> Am 14.08.19 um 17:12 schrieb Joel Serrano:
>> Hello, 
>>
>> Simple doubt regarding dispatcher module when you have
>> the ds_default_socket() modparam set:
>>
>> Say you have 2 kamailio boxes, active + passive, one shared IP
>> via keepalived On the actrive node, you have
>> dispatcher.send_ping to 1, and the passive has it set to 0.
>>
>> So far OK, only the active box is sending out probing OPTIONS
>> requests, and it's using as outbound socket the virtual IP out of
>> all the available ones (as defined with ds_default_socket() modparam)
>>
>> I have seen in a capture that the outbound OPTIONS look like:
>>
>> OPTIONS sip:190.14.203.219:5060  SIP/2.0
>> Via: SIP/2.0/UDP
>> A.B.C.180;branch=z9hG4bK7abb.6fbaccc1.0
>> To: http://190.14.203.219:5060>>
>> From: 
>> ;tag=e2bdd495733c59fd91487a137fccad4e-8a73
>> CSeq: 10 OPTIONS
>> Call-ID: 2019f8491329c382-31419@A.B.C.181
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> A.B.C.180 -> VRRP
>> A.B.C.181 -> Physical IP of the box
>>
>> My doubt is, shouldn't ds_default_socket also update the IP used
>> to generate the Call-ID used in the OPTIONS request? Currently, I
>> believe it's using the first defined listen= IP from config
>> script as I switched the order the listen= are defined and I get
>> the desired result:
>>
>> OPTIONS sip:186.188.220.174:5060 
>> SIP/2.0
>> Via: SIP/2.0/UDP
>> A.B.C.180;branch=z9hG4bKc97e.a8d9e1c2.0
>> To: http://186.188.220.174:5060>>
>> From: 
>> ;tag=5e2e1773f812f6a7e4085c5d036e29d8-d323
>> CSeq: 10 OPTIONS
>> Call-ID: 7d9a92c218fc1ba0-32111@A.B.C.180
>> Max-Forwards: 70
>> Content-Length: 0
>>
>>
>> Is this expected?
>>
>> Cheers, 
>> Joel.
>>
>>
>>
>>
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>
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> Kamailio services - https://skalatan.de/services
>
>
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Re: [SR-Users] Question regarding Dispatcher + ds_default_socket() modparam + Call-ID generation

[Joel Serrano]

Hello Henning,

No concerns at all!! As you say, the Call-ID can really say whatever...
The only concern could/would be in the security topic that you are
disclosing potential sensible information about your infrastructure
blablabla... but that can be solved just by changing the listen= order so
even that wouldn't be a problem.. In reality I was just curious so I
thought I'd ask :)

Thanks!!
Joel.



On Wed, Aug 14, 2019 at 9:32 AM Henning Westerholt  wrote:

> Hello Joel,
>
> funny - I just had this discussion about the same topic some days ago.
>
> In the end this is "only" the call-id, the IP should not be used to to
> routing descisions etc.. Do you have some more concerns about this?
>
> I think as well it just uses the first IP. I think at the moment the
> call-id is generated internally from tm, but this could be of course
> changed for the module with some coding/extension.
>
> Cheers,
>
> Henning
> Am 14.08.19 um 17:12 schrieb Joel Serrano:
>
> Hello,
>
> Simple doubt regarding dispatcher module when you have
> the ds_default_socket() modparam set:
>
> Say you have 2 kamailio boxes, active + passive, one shared IP via
> keepalived On the actrive node, you have dispatcher.send_ping to 1, and
> the passive has it set to 0.
>
> So far OK, only the active box is sending out probing OPTIONS requests,
> and it's using as outbound socket the virtual IP out of all the available
> ones (as defined with ds_default_socket() modparam)
>
> I have seen in a capture that the outbound OPTIONS look like:
>
> OPTIONS sip:190.14.203.219:5060 SIP/2.0
> Via: SIP/2.0/UDP A.B.C.180
> ;branch=z9hG4bK7abb.6fbaccc1.0
> To: 
> From:  
> ;tag=e2bdd495733c59fd91487a137fccad4e-8a73
> CSeq: 10 OPTIONS
> Call-ID: 2019f8491329c382-31419@A.B.C.181
> Max-Forwards: 70
> Content-Length: 0
>
> A.B.C.180 -> VRRP
> A.B.C.181 -> Physical IP of the box
>
> My doubt is, shouldn't ds_default_socket also update the IP used to
> generate the Call-ID used in the OPTIONS request? Currently, I believe it's
> using the first defined listen= IP from config script as I switched the
> order the listen= are defined and I get the desired result:
>
> OPTIONS sip:186.188.220.174:5060 SIP/2.0
> Via: SIP/2.0/UDP A.B.C.180
> ;branch=z9hG4bKc97e.a8d9e1c2.0
> To: 
> From:  
> ;tag=5e2e1773f812f6a7e4085c5d036e29d8-d323
> CSeq: 10 OPTIONS
> Call-ID: 7d9a92c218fc1ba0-32111@A.B.C.180
> Max-Forwards: 70
> Content-Length: 0
>
>
> Is this expected?
>
> Cheers,
> Joel.
>
>
>
>
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Re: [SR-Users] Question regarding Dispatcher + ds_default_socket() modparam + Call-ID generation

[Henning Westerholt]

Hello Joel,

funny - I just had this discussion about the same topic some days ago.

In the end this is "only" the call-id, the IP should not be used to to routing 
descisions etc.. Do you have some more concerns about this?

I think as well it just uses the first IP. I think at the moment the call-id is 
generated internally from tm, but this could be of course changed for the 
module with some coding/extension.

Cheers,

Henning

Am 14.08.19 um 17:12 schrieb Joel Serrano:
Hello,

Simple doubt regarding dispatcher module when you have the ds_default_socket() 
modparam set:

Say you have 2 kamailio boxes, active + passive, one shared IP via 
keepalived On the actrive node, you have dispatcher.send_ping to 1, and the 
passive has it set to 0.

So far OK, only the active box is sending out probing OPTIONS requests, and 
it's using as outbound socket the virtual IP out of all the available ones (as 
defined with ds_default_socket() modparam)

I have seen in a capture that the outbound OPTIONS look like:

OPTIONS sip:190.14.203.219:5060 SIP/2.0
Via: SIP/2.0/UDP A.B.C.180;branch=z9hG4bK7abb.6fbaccc1.0
To: http://190.14.203.219:5060>>
From: 
;tag=e2bdd495733c59fd91487a137fccad4e-8a73
CSeq: 10 OPTIONS
Call-ID: 2019f8491329c382-31419@A.B.C.181
Max-Forwards: 70
Content-Length: 0

A.B.C.180 -> VRRP
A.B.C.181 -> Physical IP of the box

My doubt is, shouldn't ds_default_socket also update the IP used to generate 
the Call-ID used in the OPTIONS request? Currently, I believe it's using the 
first defined listen= IP from config script as I switched the order the listen= 
are defined and I get the desired result:

OPTIONS sip:186.188.220.174:5060 SIP/2.0
Via: SIP/2.0/UDP A.B.C.180;branch=z9hG4bKc97e.a8d9e1c2.0
To: http://186.188.220.174:5060>>
From: 
;tag=5e2e1773f812f6a7e4085c5d036e29d8-d323
CSeq: 10 OPTIONS
Call-ID: 7d9a92c218fc1ba0-32111@A.B.C.180
Max-Forwards: 70
Content-Length: 0


Is this expected?

Cheers,
Joel.






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[SR-Users] Question regarding Dispatcher + ds_default_socket() modparam + Call-ID generation

[Joel Serrano]

Hello,

Simple doubt regarding dispatcher module when you have
the ds_default_socket() modparam set:

Say you have 2 kamailio boxes, active + passive, one shared IP via
keepalived On the actrive node, you have dispatcher.send_ping to 1, and
the passive has it set to 0.

So far OK, only the active box is sending out probing OPTIONS requests, and
it's using as outbound socket the virtual IP out of all the available ones
(as defined with ds_default_socket() modparam)

I have seen in a capture that the outbound OPTIONS look like:

OPTIONS sip:190.14.203.219:5060 SIP/2.0
Via: SIP/2.0/UDP A.B.C.180
;branch=z9hG4bK7abb.6fbaccc1.0
To: 
From: ;tag=e2bdd495733c59fd91487a137fccad4e-8a73
CSeq: 10 OPTIONS
Call-ID: 2019f8491329c382-31419@A.B.C.181
Max-Forwards: 70
Content-Length: 0

A.B.C.180 -> VRRP
A.B.C.181 -> Physical IP of the box

My doubt is, shouldn't ds_default_socket also update the IP used to
generate the Call-ID used in the OPTIONS request? Currently, I believe it's
using the first defined listen= IP from config script as I switched the
order the listen= are defined and I get the desired result:

OPTIONS sip:186.188.220.174:5060 SIP/2.0
Via: SIP/2.0/UDP A.B.C.180
;branch=z9hG4bKc97e.a8d9e1c2.0
To: 
From: ;tag=5e2e1773f812f6a7e4085c5d036e29d8-d323
CSeq: 10 OPTIONS
Call-ID: 7d9a92c218fc1ba0-32111@A.B.C.180
Max-Forwards: 70
Content-Length: 0


Is this expected?

Cheers,
Joel.
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Re: [SR-Users] question about reply route

[Iván Aponte]

Thanks Joel and Karsten both ideas worked.

On Sat, Jun 29, 2019 at 10:13 AM Karsten Horsmann  wrote:
>
> Hi,
>
> Or you could configure
> one listen  advertise pubip
> And a second listen internal ip (could be from the same private subnet) and 
> you point your Asterisk to the private ip.
>
>
> Cheers
> Karsten
>
>
> Joel Serrano  schrieb am Sa., 29. Juni 2019, 06:51:
>>
>> Using the “advertise” option in the listen= param will cause the route/via 
>> headers to use whatever you configure there regardless of what the actual 
>> interface IP is.
>>
>> One option is for you enable a second port for communication with asterisk, 
>> and leave 5060 with the advertise for your trunk...
>>
>>
>> On Fri, Jun 28, 2019 at 19:28 Iván Aponte  wrote:
>>>
>>> Hello,
>>>
>>> I have a question about record route and via headers. My scenario is
>>> the following:
>>>
>>> [asterisk]  [kam]  [trunk]
>>> 10.142.0.27 10.142.0.6200.x.x.x
>>>
>>> I have an asterisk an a kamailio in a private ip network
>>> (10.142.0.0/24). Kamailio has the address 10.142.0.6 but is also
>>> natted 1:1 to a public ip (35.0.0.6).
>>>
>>> My problem is that when the invite are replied the asterisk replies to
>>>  the external  ip (35.0.0.6) instead of the internal ip (even though
>>> the kamailio replies goes through the internal network) . I tried
>>> adding  a reply route to change the advertised address to no avail.
>>> any advice would be appreciated.
>>>
>>>
>>> this is my conf:
>>>
>>> LISTEN="listen=udp:0.0.0.0:5060 advertise 35.0.0.6:5060"
>>> disable_tcp=yes
>>> loadmodule "jsonrpcs.so"
>>> loadmodule "kex.so"
>>> loadmodule "corex.so"
>>> loadmodule "tm.so"
>>> loadmodule "tmx.so"
>>> loadmodule "sl.so"
>>> loadmodule "rr.so"
>>> loadmodule "pv.so"
>>> loadmodule "maxfwd.so"
>>> loadmodule "usrloc.so"
>>> loadmodule "textops.so"
>>> loadmodule "siputils.so"
>>> loadmodule "xlog.so"
>>> loadmodule "sanity.so"
>>> loadmodule "ctl.so"
>>> loadmodule "cfg_rpc.so"
>>> loadmodule "acc.so"
>>> loadmodule "counters.so"
>>> loadmodule "permissions.so"
>>> loadmodule "ipops.so"
>>>
>>> modparam("jsonrpcs", "pretty_format", 1)
>>> modparam("tm", "failure_reply_mode", 3)
>>> modparam("tm", "fr_timer", 3)
>>> modparam("tm", "fr_inv_timer", 12)
>>>
>>> modparam("rr", "enable_full_lr", 0)
>>> modparam("rr", "append_fromtag", 0)
>>>
>>> reply_route {
>>>xinfo("source $si");
>>> if(!is_in_subnet($si, "10.0.0.0/8")) {
>>>  set_advertised_address("10.142.0.6");
>>>}
>>>else {
>>>  xinfo("not in subnet");
>>>}
>>> }
>>>
>>>
>>> request_route {
>>>
>>> # per request initial checks
>>> route(REQINIT);
>>> # CANCEL processing
>>> if (is_method("CANCEL")) {
>>> if (t_check_trans()) {
>>> route(RELAY);
>>> }
>>> exit;
>>> }
>>>
>>> # handle retransmissions
>>> if (!is_method("ACK")) {
>>> if(t_precheck_trans()) {
>>> t_check_trans();
>>> exit;
>>> }
>>> t_check_trans();
>>> }
>>>
>>> # handle requests within SIP dialogs
>>> route(WITHINDLG);
>>>
>>> ### only initial requests (no To tag)
>>>
>>> # authentication
>>> #route(AUTH);
>>>
>>> # record routing for dialog forming requests (in case they are routed)
>>> # - remove preloaded route headers
>>> remove_hf("Route");
>>> if (is_method("INVITE|SUBSCRIBE")) {
>>> record_route();
>>> }
>>>
>>>
>>> # dispatch requests to foreign domains
>>> route(SIPOUT);
>>>
>>> if ($rU==$null) {
>>> # request with no Username in RURI
>>> sl_send_reply("484","Address Incomplete");
>>> exit;
>>> }
>>>
>>> }
>>>
>>> # Wrapper for relaying requests
>>> route[RELAY] {
>>> if (!t_relay()) {
>>> sl_reply_error();
>>> }
>>> exit;
>>> }
>>>
>>> # Per SIP request initial checks
>>> route[REQINIT] {
>>> if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
>>> # silent drop for scanners - uncomment next line if want to reply
>>> # sl_send_reply("200", "OK");
>>> exit;
>>> }
>>>
>>> if (!mf_process_maxfwd_header("10")) {
>>> sl_send_reply("483","Too Many Hops");
>>> exit;
>>> }
>>>
>>> if(is_method("OPTIONS") && uri==myself && $rU==$null) {
>>> sl_send_reply("200","Keepalive");
>>> exit;
>>> }
>>>
>>> if(!sanity_check("1511", "7")) {
>>> xlog("Malformed SIP message from $si:$sp\n");
>>> exit;
>>> }
>>> }
>>>
>>> # Handle requests within SIP dialogs
>>> route[WITHINDLG] {
>>> if (!has_totag()) return;
>>>
>>> # sequential request withing a dialog should
>>> # take the path determined by record-routing
>>> if (loose_route()) {
>>> if ( is_method("NOTIFY") ) {
>>> record_route();
>>> }
>>> route(RELAY);
>>> exit;
>>> }
>>>
>>> if (is_method("SUBSCRIBE") && uri == myself) {
>>> send_reply("405", "Method Not Allowed");
>>> exit;
>>> }
>>> if ( is_method("ACK") ) {
>>> if ( t_check_trans() ) {
>>> # no loose-route, but stateful ACK;
>>> # must be an ACK after a 487
>>> # or e.g. 404 from upstream server
>>> route(RELAY);
>>> exit;
>>> } else {
>>> # ACK without matching transaction ... ignore and discard
>>> exit;
>>> }
>>> }
>>> sl_send_reply("404","Not here");
>>> exit;
>>> }
>>> route[SIP

Re: [SR-Users] question about reply route

[Karsten Horsmann]

Hi,

Or you could configure
one listen  advertise pubip
And a second listen internal ip (could be from the same private subnet) and
you point your Asterisk to the private ip.


Cheers
Karsten


Joel Serrano  schrieb am Sa., 29. Juni 2019, 06:51:

> Using the “advertise” option in the listen= param will cause the route/via
> headers to use whatever you configure there regardless of what the actual
> interface IP is.
>
> One option is for you enable a second port for communication with
> asterisk, and leave 5060 with the advertise for your trunk...
>
>
> On Fri, Jun 28, 2019 at 19:28 Iván Aponte  wrote:
>
>> Hello,
>>
>> I have a question about record route and via headers. My scenario is
>> the following:
>>
>> [asterisk]  [kam]  [trunk]
>> 10.142.0.27 10.142.0.6200.x.x.x
>>
>> I have an asterisk an a kamailio in a private ip network
>> (10.142.0.0/24). Kamailio has the address 10.142.0.6 but is also
>> natted 1:1 to a public ip (35.0.0.6).
>>
>> My problem is that when the invite are replied the asterisk replies to
>>  the external  ip (35.0.0.6) instead of the internal ip (even though
>> the kamailio replies goes through the internal network) . I tried
>> adding  a reply route to change the advertised address to no avail.
>> any advice would be appreciated.
>>
>>
>> this is my conf:
>>
>> LISTEN="listen=udp:0.0.0.0:5060 advertise 35.0.0.6:5060"
>> disable_tcp=yes
>> loadmodule "jsonrpcs.so"
>> loadmodule "kex.so"
>> loadmodule "corex.so"
>> loadmodule "tm.so"
>> loadmodule "tmx.so"
>> loadmodule "sl.so"
>> loadmodule "rr.so"
>> loadmodule "pv.so"
>> loadmodule "maxfwd.so"
>> loadmodule "usrloc.so"
>> loadmodule "textops.so"
>> loadmodule "siputils.so"
>> loadmodule "xlog.so"
>> loadmodule "sanity.so"
>> loadmodule "ctl.so"
>> loadmodule "cfg_rpc.so"
>> loadmodule "acc.so"
>> loadmodule "counters.so"
>> loadmodule "permissions.so"
>> loadmodule "ipops.so"
>>
>> modparam("jsonrpcs", "pretty_format", 1)
>> modparam("tm", "failure_reply_mode", 3)
>> modparam("tm", "fr_timer", 3)
>> modparam("tm", "fr_inv_timer", 12)
>>
>> modparam("rr", "enable_full_lr", 0)
>> modparam("rr", "append_fromtag", 0)
>>
>> reply_route {
>>xinfo("source $si");
>> if(!is_in_subnet($si, "10.0.0.0/8")) {
>>  set_advertised_address("10.142.0.6");
>>}
>>else {
>>  xinfo("not in subnet");
>>}
>> }
>>
>>
>> request_route {
>>
>> # per request initial checks
>> route(REQINIT);
>> # CANCEL processing
>> if (is_method("CANCEL")) {
>> if (t_check_trans()) {
>> route(RELAY);
>> }
>> exit;
>> }
>>
>> # handle retransmissions
>> if (!is_method("ACK")) {
>> if(t_precheck_trans()) {
>> t_check_trans();
>> exit;
>> }
>> t_check_trans();
>> }
>>
>> # handle requests within SIP dialogs
>> route(WITHINDLG);
>>
>> ### only initial requests (no To tag)
>>
>> # authentication
>> #route(AUTH);
>>
>> # record routing for dialog forming requests (in case they are routed)
>> # - remove preloaded route headers
>> remove_hf("Route");
>> if (is_method("INVITE|SUBSCRIBE")) {
>> record_route();
>> }
>>
>>
>> # dispatch requests to foreign domains
>> route(SIPOUT);
>>
>> if ($rU==$null) {
>> # request with no Username in RURI
>> sl_send_reply("484","Address Incomplete");
>> exit;
>> }
>>
>> }
>>
>> # Wrapper for relaying requests
>> route[RELAY] {
>> if (!t_relay()) {
>> sl_reply_error();
>> }
>> exit;
>> }
>>
>> # Per SIP request initial checks
>> route[REQINIT] {
>> if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
>> # silent drop for scanners - uncomment next line if want to reply
>> # sl_send_reply("200", "OK");
>> exit;
>> }
>>
>> if (!mf_process_maxfwd_header("10")) {
>> sl_send_reply("483","Too Many Hops");
>> exit;
>> }
>>
>> if(is_method("OPTIONS") && uri==myself && $rU==$null) {
>> sl_send_reply("200","Keepalive");
>> exit;
>> }
>>
>> if(!sanity_check("1511", "7")) {
>> xlog("Malformed SIP message from $si:$sp\n");
>> exit;
>> }
>> }
>>
>> # Handle requests within SIP dialogs
>> route[WITHINDLG] {
>> if (!has_totag()) return;
>>
>> # sequential request withing a dialog should
>> # take the path determined by record-routing
>> if (loose_route()) {
>> if ( is_method("NOTIFY") ) {
>> record_route();
>> }
>> route(RELAY);
>> exit;
>> }
>>
>> if (is_method("SUBSCRIBE") && uri == myself) {
>> send_reply("405", "Method Not Allowed");
>> exit;
>> }
>> if ( is_method("ACK") ) {
>> if ( t_check_trans() ) {
>> # no loose-route, but stateful ACK;
>> # must be an ACK after a 487
>> # or e.g. 404 from upstream server
>> route(RELAY);
>> exit;
>> } else {
>> # ACK without matching transaction ... ignore and discard
>> exit;
>> }
>> }
>> sl_send_reply("404","Not here");
>> exit;
>> }
>> route[SIPOUT] {
>> if (uri==myself) return;
>> append_hf("P-hint: outbound\r\n");
>> route(RELAY);
>> exit;
>> }
>>
>>
>>
>>
>> --
>> Iván Aponte
>> Office: +58(212)9923193
>> Mobile: +58(412)2774713
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> h

Re: [SR-Users] question about reply route

[Joel Serrano]

Using the “advertise” option in the listen= param will cause the route/via
headers to use whatever you configure there regardless of what the actual
interface IP is.

One option is for you enable a second port for communication with asterisk,
and leave 5060 with the advertise for your trunk...


On Fri, Jun 28, 2019 at 19:28 Iván Aponte  wrote:

> Hello,
>
> I have a question about record route and via headers. My scenario is
> the following:
>
> [asterisk]  [kam]  [trunk]
> 10.142.0.27 10.142.0.6200.x.x.x
>
> I have an asterisk an a kamailio in a private ip network
> (10.142.0.0/24). Kamailio has the address 10.142.0.6 but is also
> natted 1:1 to a public ip (35.0.0.6).
>
> My problem is that when the invite are replied the asterisk replies to
>  the external  ip (35.0.0.6) instead of the internal ip (even though
> the kamailio replies goes through the internal network) . I tried
> adding  a reply route to change the advertised address to no avail.
> any advice would be appreciated.
>
>
> this is my conf:
>
> LISTEN="listen=udp:0.0.0.0:5060 advertise 35.0.0.6:5060"
> disable_tcp=yes
> loadmodule "jsonrpcs.so"
> loadmodule "kex.so"
> loadmodule "corex.so"
> loadmodule "tm.so"
> loadmodule "tmx.so"
> loadmodule "sl.so"
> loadmodule "rr.so"
> loadmodule "pv.so"
> loadmodule "maxfwd.so"
> loadmodule "usrloc.so"
> loadmodule "textops.so"
> loadmodule "siputils.so"
> loadmodule "xlog.so"
> loadmodule "sanity.so"
> loadmodule "ctl.so"
> loadmodule "cfg_rpc.so"
> loadmodule "acc.so"
> loadmodule "counters.so"
> loadmodule "permissions.so"
> loadmodule "ipops.so"
>
> modparam("jsonrpcs", "pretty_format", 1)
> modparam("tm", "failure_reply_mode", 3)
> modparam("tm", "fr_timer", 3)
> modparam("tm", "fr_inv_timer", 12)
>
> modparam("rr", "enable_full_lr", 0)
> modparam("rr", "append_fromtag", 0)
>
> reply_route {
>xinfo("source $si");
> if(!is_in_subnet($si, "10.0.0.0/8")) {
>  set_advertised_address("10.142.0.6");
>}
>else {
>  xinfo("not in subnet");
>}
> }
>
>
> request_route {
>
> # per request initial checks
> route(REQINIT);
> # CANCEL processing
> if (is_method("CANCEL")) {
> if (t_check_trans()) {
> route(RELAY);
> }
> exit;
> }
>
> # handle retransmissions
> if (!is_method("ACK")) {
> if(t_precheck_trans()) {
> t_check_trans();
> exit;
> }
> t_check_trans();
> }
>
> # handle requests within SIP dialogs
> route(WITHINDLG);
>
> ### only initial requests (no To tag)
>
> # authentication
> #route(AUTH);
>
> # record routing for dialog forming requests (in case they are routed)
> # - remove preloaded route headers
> remove_hf("Route");
> if (is_method("INVITE|SUBSCRIBE")) {
> record_route();
> }
>
>
> # dispatch requests to foreign domains
> route(SIPOUT);
>
> if ($rU==$null) {
> # request with no Username in RURI
> sl_send_reply("484","Address Incomplete");
> exit;
> }
>
> }
>
> # Wrapper for relaying requests
> route[RELAY] {
> if (!t_relay()) {
> sl_reply_error();
> }
> exit;
> }
>
> # Per SIP request initial checks
> route[REQINIT] {
> if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
> # silent drop for scanners - uncomment next line if want to reply
> # sl_send_reply("200", "OK");
> exit;
> }
>
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
> }
>
> if(is_method("OPTIONS") && uri==myself && $rU==$null) {
> sl_send_reply("200","Keepalive");
> exit;
> }
>
> if(!sanity_check("1511", "7")) {
> xlog("Malformed SIP message from $si:$sp\n");
> exit;
> }
> }
>
> # Handle requests within SIP dialogs
> route[WITHINDLG] {
> if (!has_totag()) return;
>
> # sequential request withing a dialog should
> # take the path determined by record-routing
> if (loose_route()) {
> if ( is_method("NOTIFY") ) {
> record_route();
> }
> route(RELAY);
> exit;
> }
>
> if (is_method("SUBSCRIBE") && uri == myself) {
> send_reply("405", "Method Not Allowed");
> exit;
> }
> if ( is_method("ACK") ) {
> if ( t_check_trans() ) {
> # no loose-route, but stateful ACK;
> # must be an ACK after a 487
> # or e.g. 404 from upstream server
> route(RELAY);
> exit;
> } else {
> # ACK without matching transaction ... ignore and discard
> exit;
> }
> }
> sl_send_reply("404","Not here");
> exit;
> }
> route[SIPOUT] {
> if (uri==myself) return;
> append_hf("P-hint: outbound\r\n");
> route(RELAY);
> exit;
> }
>
>
>
>
> --
> Iván Aponte
> Office: +58(212)9923193
> Mobile: +58(412)2774713
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
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[SR-Users] question about reply route

[Iván Aponte]

Hello,

I have a question about record route and via headers. My scenario is
the following:

[asterisk]  [kam]  [trunk]
10.142.0.27 10.142.0.6200.x.x.x

I have an asterisk an a kamailio in a private ip network
(10.142.0.0/24). Kamailio has the address 10.142.0.6 but is also
natted 1:1 to a public ip (35.0.0.6).

My problem is that when the invite are replied the asterisk replies to
 the external  ip (35.0.0.6) instead of the internal ip (even though
the kamailio replies goes through the internal network) . I tried
adding  a reply route to change the advertised address to no avail.
any advice would be appreciated.


this is my conf:

LISTEN="listen=udp:0.0.0.0:5060 advertise 35.0.0.6:5060"
disable_tcp=yes
loadmodule "jsonrpcs.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.so"
loadmodule "permissions.so"
loadmodule "ipops.so"

modparam("jsonrpcs", "pretty_format", 1)
modparam("tm", "failure_reply_mode", 3)
modparam("tm", "fr_timer", 3)
modparam("tm", "fr_inv_timer", 12)

modparam("rr", "enable_full_lr", 0)
modparam("rr", "append_fromtag", 0)

reply_route {
   xinfo("source $si");
if(!is_in_subnet($si, "10.0.0.0/8")) {
 set_advertised_address("10.142.0.6");
   }
   else {
 xinfo("not in subnet");
   }
}


request_route {

# per request initial checks
route(REQINIT);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}

# handle retransmissions
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}

# handle requests within SIP dialogs
route(WITHINDLG);

### only initial requests (no To tag)

# authentication
#route(AUTH);

# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")) {
record_route();
}


# dispatch requests to foreign domains
route(SIPOUT);

if ($rU==$null) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}

}

# Wrapper for relaying requests
route[RELAY] {
if (!t_relay()) {
sl_reply_error();
}
exit;
}

# Per SIP request initial checks
route[REQINIT] {
if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
# silent drop for scanners - uncomment next line if want to reply
# sl_send_reply("200", "OK");
exit;
}

if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}

if(is_method("OPTIONS") && uri==myself && $rU==$null) {
sl_send_reply("200","Keepalive");
exit;
}

if(!sanity_check("1511", "7")) {
xlog("Malformed SIP message from $si:$sp\n");
exit;
}
}

# Handle requests within SIP dialogs
route[WITHINDLG] {
if (!has_totag()) return;

# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if ( is_method("NOTIFY") ) {
record_route();
}
route(RELAY);
exit;
}

if (is_method("SUBSCRIBE") && uri == myself) {
send_reply("405", "Method Not Allowed");
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
route(RELAY);
exit;
} else {
# ACK without matching transaction ... ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
exit;
}
route[SIPOUT] {
if (uri==myself) return;
append_hf("P-hint: outbound\r\n");
route(RELAY);
exit;
}




-- 
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Office: +58(212)9923193
Mobile: +58(412)2774713

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Re: [SR-Users] Question about registrar behavior

[Володимир Іванець]

Hello Henning,

I'm preparing Kamailio configuration that uses save() responce codes for
additional actions so email from Lars brought my attention. This is not a
problem since as Lars mentioned I can check expires value. I just was
curious if this is a correct behavior or not.

Thank you!

ср, 26 черв. 2019 о 23:36 Henning Westerholt  пише:

> Hello,
>
> just briefly looked into the code, but I think the return value is like
> this for the following reasons:
>
> - function update_contacts(..) will skip if a contact is not found and a
> expires=0 is given (it could be also e.g. that the Contact just expired a
> few seconds ago in Kamailio)
>
> - this function will return 0, means success
>
> - the calling function save(..) will then return 1 as the default return
> value
>
> The main question to change this would be how to differentiate between a
> the valid case (just expired in Kamailio) from the other case (simply not
> registered at all). What issues do you experience because of this behaviour?
>
> Cheers,
>
> Henning
> Am 26.06.19 um 16:25 schrieb Володимир Іванець:
>
> Hello!
>
> I've just tested this on Kamailio v. 5.3.0-dev6 with *modparam("usrloc",
> "db_mode", 0)* setting. save() return code was 1 too. I'm also interested
> if this behavior is by design.
>
> Thanks.
>
> ср, 26 черв. 2019 о 11:34 Lars Olsson  пише:
>
>> Hi,
>>
>> I have found a behavior in the registrar module that I do have a question
>> about. Is the current behavior correct and wanted?
>>
>> Using the save() method in the script I see the following:
>>
>>
>>- Processing a register request for a user gives return code 1 ( or 2
>>)
>>- Processing a unregister request (expires=0) for registered user
>>gives return code 3
>>- Processing a unregister request (expires=0) for a user which is NOT
>>registered gives return code 1. Why?
>>
>> What is the reason behind this?
>> No database entry is added which is expected.
>>
>> Test performed on 5.1.4, using DB mode 3.
>>
>> For handling a late unregister request ( where registration has already
>> expired) return code does not reflect the the action.
>> I assume that manually checking $expires(max) is the option to go then if
>> I want to detect the unregister request or?
>>
>> Cheers,
>> Lars
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
> ___
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>
> --
> Henning Westerholt - https://skalatan.de/blog/
> Kamailio services - https://skalatan.de/services
>
>
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Re: [SR-Users] Question about registrar behavior

[Henning Westerholt]

Hello,

just briefly looked into the code, but I think the return value is like this 
for the following reasons:

- function update_contacts(..) will skip if a contact is not found and a 
expires=0 is given (it could be also e.g. that the Contact just expired a few 
seconds ago in Kamailio)

- this function will return 0, means success

- the calling function save(..) will then return 1 as the default return value

The main question to change this would be how to differentiate between a the 
valid case (just expired in Kamailio) from the other case (simply not 
registered at all). What issues do you experience because of this behaviour?

Cheers,

Henning

Am 26.06.19 um 16:25 schrieb Володимир Іванець:
Hello!

I've just tested this on Kamailio v. 5.3.0-dev6 with modparam("usrloc", 
"db_mode", 0) setting. save() return code was 1 too. I'm also interested if 
this behavior is by design.

Thanks.

ср, 26 черв. 2019 о 11:34 Lars Olsson 
mailto:lars.ols...@optimobile.se>> пише:
Hi,

I have found a behavior in the registrar module that I do have a question 
about. Is the current behavior correct and wanted?

Using the save() method in the script I see the following:


  *   Processing a register request for a user gives return code 1 ( or 2 )
  *   Processing a unregister request (expires=0) for registered user gives 
return code 3
  *   Processing a unregister request (expires=0) for a user which is NOT 
registered gives return code 1. Why?

What is the reason behind this?
No database entry is added which is expected.

Test performed on 5.1.4, using DB mode 3.

For handling a late unregister request ( where registration has already 
expired) return code does not reflect the the action.
I assume that manually checking $expires(max) is the option to go then if I 
want to detect the unregister request or?

Cheers,
Lars

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--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://skalatan.de/services
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Re: [SR-Users] Question about registrar behavior

[Володимир Іванець]

Hello!

I've just tested this on Kamailio v. 5.3.0-dev6 with *modparam("usrloc",
"db_mode", 0)* setting. save() return code was 1 too. I'm also interested
if this behavior is by design.

Thanks.

ср, 26 черв. 2019 о 11:34 Lars Olsson  пише:

> Hi,
>
> I have found a behavior in the registrar module that I do have a question
> about. Is the current behavior correct and wanted?
>
> Using the save() method in the script I see the following:
>
>
>- Processing a register request for a user gives return code 1 ( or 2 )
>- Processing a unregister request (expires=0) for registered user
>gives return code 3
>- Processing a unregister request (expires=0) for a user which is NOT
>registered gives return code 1. Why?
>
> What is the reason behind this?
> No database entry is added which is expected.
>
> Test performed on 5.1.4, using DB mode 3.
>
> For handling a late unregister request ( where registration has already
> expired) return code does not reflect the the action.
> I assume that manually checking $expires(max) is the option to go then if
> I want to detect the unregister request or?
>
> Cheers,
> Lars
>
> ___
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> sr-users@lists.kamailio.org
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[SR-Users] Question about registrar behavior

[Lars Olsson]

Hi,

I have found a behavior in the registrar module that I do have a question 
about. Is the current behavior correct and wanted?

Using the save() method in the script I see the following:


  *   Processing a register request for a user gives return code 1 ( or 2 )
  *   Processing a unregister request (expires=0) for registered user gives 
return code 3
  *   Processing a unregister request (expires=0) for a user which is NOT 
registered gives return code 1. Why?

What is the reason behind this?
No database entry is added which is expected.

Test performed on 5.1.4, using DB mode 3.

For handling a late unregister request ( where registration has already 
expired) return code does not reflect the the action.
I assume that manually checking $expires(max) is the option to go then if I 
want to detect the unregister request or?

Cheers,
Lars

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