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!
>>
>> __
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>>   * sr-users@lists.kamailio.org
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>> 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)
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>
>
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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!
>>
>> __
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>> the 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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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!
>
> __
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> --
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> 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!
>
> __
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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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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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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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> www.linkedin.com/in/miconda
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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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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

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
[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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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

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 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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>> the sender!
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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
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> the sender!
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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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> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
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>
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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
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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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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  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 " 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]

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

Outlook voor Android 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" 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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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
>>>
>> ___
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>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
> ___
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>
> --
> 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
>>>
>> ___
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>> 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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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
>>
>>
>> ___
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>> sr-users@lists.kamailio.org 
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
>> 
>
> -- 
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> 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
>>
>> ___
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>> 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
>
> ___
> 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
>
> ___
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> 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,

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@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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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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achen@ fuze.com

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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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achen@ fuze.com

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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.

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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?


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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?


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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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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

--
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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.

--
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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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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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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

-- 
Henning Westerholt – 
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 – 
https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgilawa.com&data=02%7C01%7Cf.floimair%40commend.com%7Cb26fa49c351b49ce55e408d7799068cc%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637111533448085251&sdata=0wcQT0o9TUXpPziAaqHJw1%2F4kV6sq%2F2PBGomIUrWELg%3D&reserved=0
 

-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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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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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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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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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> -- 
> Henning Westerholt - https://skalatan.de/blog/
> 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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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
>>
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>> 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
>
> ___
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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 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
>>
>> ___
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>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
> ___
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>
> --
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> 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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Re: [SR-Users] Question on Database Version Table

[Daniel-Constantin Mierla]

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

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


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

[Alex Balashov]

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.

-- 
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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

[Daniel-Constantin Mierla]

On 18.06.19 21:50, Alex Balashov wrote:
> It's probably also worth noting that Kamailio's DB layer offers
> "unpredictable" reactions to columns which are missing and which the
> given version expects to be there. 

Where did you meet such situation? A missing but expected column should
end up in an error at the sql query layer, because kamailio modules
explicitely do select/insert/... with column names.

The only unpredictable situation with some modules may be when the
column exist, but has a different type than expected. But a missing
column should be handled because of the db query error.

Cheers,
Daniel

>
> "Unpredictable" generally means a crash, but when it happens -- the
> presence of columns isn't always validated at boot time, depends on the
> module -- is the unpredictable part... 
>
> So, modernising your DB schema for built-in module tables, including
> seldom-used tables which are nevertheless formally used to back some
> module, is important when doing upgrades.
>
> On Tue, Jun 18, 2019 at 02:46:27PM -0500, JR Richardson wrote:
>
>> Yep
>> version_table
>> Set the name of the table holding the table version. Useful if the
>> proxy is sharing a database within a project and during upgrades.
>> Default value is “version”.
>> Example of usage:
>> version_table="version44"
>>
>> That's what I was looking for, perfect!
>>
>> Yes, I added accountcode and notes to my table for internal use, was
>> just making notes on columns I need on the new version upgrade.
>>
>> Thanks.
>>
>> JR
>>
>>
>> On Tue, Jun 18, 2019 at 2:17 PM Daniel-Constantin Mierla
>>  wrote:
>>> Hello,
>>>
>>> like Alex said, you can have two version tables, one for each versions
>>> of kamailio you play with and use in config the core parameter to set
>>> the name of that table.
>>>
>>> I am actually writing to say that address table never had accountcode
>>> and notes columns. Likely they were added in your deployment for other
>>> purposes or either the default column names were changed. The default
>>> column names are listed at:
>>>
>>>   -
>>> https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-address
>>>
>>> These were the same for rather long time, iirc the recent version number
>>> changed mainly due to with modifications done to the size, not due to
>>> new columns.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 18.06.19 20:50, JR Richardson wrote:
 Hey All,

 I'm doing some upgrades and noticed database table version numbers
 have increased.

 database table adjustments:
 | address | production ver 3 | upgrade 5.2 ver 6 | need to add
 accountcode and notes column
 | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr 
 column
 | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add
 ruri_pattern and priority column

 I run mysql master/slave so I'm concerned about the master database
 not having the correct version numbers or can I just update the
 version number in the table when I do the upgrade of the kamailio
 versions?

 Thanks.

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

[Alex Balashov]

It's probably also worth noting that Kamailio's DB layer offers
"unpredictable" reactions to columns which are missing and which the
given version expects to be there. 

"Unpredictable" generally means a crash, but when it happens -- the
presence of columns isn't always validated at boot time, depends on the
module -- is the unpredictable part... 

So, modernising your DB schema for built-in module tables, including
seldom-used tables which are nevertheless formally used to back some
module, is important when doing upgrades.

On Tue, Jun 18, 2019 at 02:46:27PM -0500, JR Richardson wrote:

> Yep
> version_table
> Set the name of the table holding the table version. Useful if the
> proxy is sharing a database within a project and during upgrades.
> Default value is “version”.
> Example of usage:
> version_table="version44"
> 
> That's what I was looking for, perfect!
> 
> Yes, I added accountcode and notes to my table for internal use, was
> just making notes on columns I need on the new version upgrade.
> 
> Thanks.
> 
> JR
> 
> 
> On Tue, Jun 18, 2019 at 2:17 PM Daniel-Constantin Mierla
>  wrote:
> >
> > Hello,
> >
> > like Alex said, you can have two version tables, one for each versions
> > of kamailio you play with and use in config the core parameter to set
> > the name of that table.
> >
> > I am actually writing to say that address table never had accountcode
> > and notes columns. Likely they were added in your deployment for other
> > purposes or either the default column names were changed. The default
> > column names are listed at:
> >
> >   -
> > https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-address
> >
> > These were the same for rather long time, iirc the recent version number
> > changed mainly due to with modifications done to the size, not due to
> > new columns.
> >
> > Cheers,
> > Daniel
> >
> > On 18.06.19 20:50, JR Richardson wrote:
> > > Hey All,
> > >
> > > I'm doing some upgrades and noticed database table version numbers
> > > have increased.
> > >
> > > database table adjustments:
> > > | address | production ver 3 | upgrade 5.2 ver 6 | need to add
> > > accountcode and notes column
> > > | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr 
> > > column
> > > | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add
> > > ruri_pattern and priority column
> > >
> > > I run mysql master/slave so I'm concerned about the master database
> > > not having the correct version numbers or can I just update the
> > > version number in the table when I do the upgrade of the kamailio
> > > versions?
> > >
> > > Thanks.
> > >
> > > JR
> 
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Re: [SR-Users] Question on Database Version Table

[JR Richardson]

Yep
version_table
Set the name of the table holding the table version. Useful if the
proxy is sharing a database within a project and during upgrades.
Default value is “version”.
Example of usage:
version_table="version44"

That's what I was looking for, perfect!

Yes, I added accountcode and notes to my table for internal use, was
just making notes on columns I need on the new version upgrade.

Thanks.

JR


On Tue, Jun 18, 2019 at 2:17 PM Daniel-Constantin Mierla
 wrote:
>
> Hello,
>
> like Alex said, you can have two version tables, one for each versions
> of kamailio you play with and use in config the core parameter to set
> the name of that table.
>
> I am actually writing to say that address table never had accountcode
> and notes columns. Likely they were added in your deployment for other
> purposes or either the default column names were changed. The default
> column names are listed at:
>
>   -
> https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-address
>
> These were the same for rather long time, iirc the recent version number
> changed mainly due to with modifications done to the size, not due to
> new columns.
>
> Cheers,
> Daniel
>
> On 18.06.19 20:50, JR Richardson wrote:
> > Hey All,
> >
> > I'm doing some upgrades and noticed database table version numbers
> > have increased.
> >
> > database table adjustments:
> > | address | production ver 3 | upgrade 5.2 ver 6 | need to add
> > accountcode and notes column
> > | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr 
> > column
> > | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add
> > ruri_pattern and priority column
> >
> > I run mysql master/slave so I'm concerned about the master database
> > not having the correct version numbers or can I just update the
> > version number in the table when I do the upgrade of the kamailio
> > versions?
> >
> > Thanks.
> >
> > JR

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

[Daniel-Constantin Mierla]

Hello,

like Alex said, you can have two version tables, one for each versions
of kamailio you play with and use in config the core parameter to set
the name of that table.

I am actually writing to say that address table never had accountcode
and notes columns. Likely they were added in your deployment for other
purposes or either the default column names were changed. The default
column names are listed at:

  -
https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-address

These were the same for rather long time, iirc the recent version number
changed mainly due to with modifications done to the size, not due to
new columns.

Cheers,
Daniel

On 18.06.19 20:50, JR Richardson wrote:
> Hey All,
>
> I'm doing some upgrades and noticed database table version numbers
> have increased.
>
> database table adjustments:
> | address | production ver 3 | upgrade 5.2 ver 6 | need to add
> accountcode and notes column
> | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr column
> | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add
> ruri_pattern and priority column
>
> I run mysql master/slave so I'm concerned about the master database
> not having the correct version numbers or can I just update the
> version number in the table when I do the upgrade of the kamailio
> versions?
>
> Thanks.
>
> JR

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


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

[Alex Balashov]

Hi JR,

A Kamailio of a given version will want the version table entries it
expects for its version, and will blow up on boot if it does not find
them. But it is safe to update them while Kamailio is running, as long
as you acknowledge that Kamailio will not start properly if it crashes
or is restarted unless the correct table versions are in place.

Best practice for upgrading in parallel - that is, any situation where
multiple Kamailio instances (e.g. active and standby) share a database
and need to both be running at the same time - is to make use of this
core config parameter:

https://www.kamailio.org/wiki/cookbooks/5.2.x/core#version_table

That way, you can make yourself a different version table with entries
appropriate for a given Kamailio release, and tell your new Kamailio
instance to use that one, while keeping the old one until you're
prepared to decommission it. Then you can rename the table to 'version',
or just keep 'version52' or whatever and have 'version_table' perennially
set in the config.

-- Alex

On Tue, Jun 18, 2019 at 01:50:35PM -0500, JR Richardson wrote:

> Hey All,
> 
> I'm doing some upgrades and noticed database table version numbers
> have increased.
> 
> database table adjustments:
> | address | production ver 3 | upgrade 5.2 ver 6 | need to add
> accountcode and notes column
> | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr column
> | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add
> ruri_pattern and priority column
> 
> I run mysql master/slave so I'm concerned about the master database
> not having the correct version numbers or can I just update the
> version number in the table when I do the upgrade of the kamailio
> versions?
> 
> Thanks.
> 
> JR
> -- 
> JR Richardson
> Engineering for the Masses
> Chasing the Azeotrope
> 
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-- 
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
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Re: [SR-Users] Question about setbflag and isbflagset usage

[Володимир Іванець]

Hello again,

I've just noticed that if new branch is created after using *t_suspend* and
*t_continue*, branch flag can not be used. Not sure if it can not be set or
checked. Doues anyone know about potential problems with such configuration?

Thanks!

вт, 28 трав. 2019 о 19:43 Володимир Іванець  пише:

> Hello Alex,
>
> Thank you for the clarification. Now I understand this behavior.
>
> пт, 24 трав. 2019 о 22:07 Alex Balashov  пише:
>
>> Branches are a creature of transactions. It follows that “branch flags”
>> are transaction-persistent flags at the branch level, rather than scoped to
>> the transaction as a whole.
>>
>> ACKs constitute a separate transaction.
>>
>> —
>> Sent from mobile, with due apologies for brevity and errors.
>>
>> On May 24, 2019, at 2:45 PM, Sergiu Pojoga  wrote:
>>
>> Then.. I don't know, lol
>>
>> But the fact that ACKs belong or not to even different transactions
>> (depending on whether positive or negative reply) must be playing a role
>> here.
>>
>> Good luck.
>>
>> On Fri, May 24, 2019, 12:11 PM Володимир Іванець, <
>> volodyaivan...@gmail.com> wrote:
>>
>>> Hello Sergiu,
>>>
>>> We allow multiple contacts to register on single account. They can have
>>> different properties.  On step 1 I'm looking for something specific which
>>> can be different for different contacts. If I understood correctly
>>> *setbflag* and *isbflagset* can set and check flags for different
>>> branches independently.
>>>
>>> Thank you!
>>>
>>> пт, 24 трав. 2019 о 18:14 Sergiu Pojoga  пише:
>>>
 I might be wrong, but isn't ACK part of a transaction? If so, you'd
 need to use *setflag*/*isflagset*

 Cheers.

 On Fri, May 24, 2019 at 10:03 AM Володимир Іванець <
 volodyaivan...@gmail.com> wrote:

> Hello everyone!
>
> I'm not sure if I correctly understood the description of functions
> *isbflagset* and *isbflagset* so once again I'm asking for help :)
>
> Here is what I do:
> 1. Asterisk sends INVITE to Kamailio. Kamailio performs lookup of
> available contacts and then few checks in branch_route. I'm adding 
> *setbflag("17",
> $T_branch_idx);* there.
> 2. Later in route[NATMANAGE] I added *if (isbflagset("17",
> $T_branch_idx)) {* and additional logic. *isbflagset* results in
> *true* (for same INVITE).
> 3. Finally one of UACs responds with 200 OK. Asterisk generates ACK
> packet that gets in route[NATMANAGE]. *isbflagset* results in *false*.
>
> So my question is: shouldn't *isbflagset* be *true* for ACK too since
> it's the same branch or I interpreted these fenctions incorrectly?
>
> Thanks a lot!
> ___
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Re: [SR-Users] Question about setbflag and isbflagset usage

[Володимир Іванець]

Hello Alex,

Thank you for the clarification. Now I understand this behavior.

пт, 24 трав. 2019 о 22:07 Alex Balashov  пише:

> Branches are a creature of transactions. It follows that “branch flags”
> are transaction-persistent flags at the branch level, rather than scoped to
> the transaction as a whole.
>
> ACKs constitute a separate transaction.
>
> —
> Sent from mobile, with due apologies for brevity and errors.
>
> On May 24, 2019, at 2:45 PM, Sergiu Pojoga  wrote:
>
> Then.. I don't know, lol
>
> But the fact that ACKs belong or not to even different transactions
> (depending on whether positive or negative reply) must be playing a role
> here.
>
> Good luck.
>
> On Fri, May 24, 2019, 12:11 PM Володимир Іванець, <
> volodyaivan...@gmail.com> wrote:
>
>> Hello Sergiu,
>>
>> We allow multiple contacts to register on single account. They can have
>> different properties.  On step 1 I'm looking for something specific which
>> can be different for different contacts. If I understood correctly
>> *setbflag* and *isbflagset* can set and check flags for different
>> branches independently.
>>
>> Thank you!
>>
>> пт, 24 трав. 2019 о 18:14 Sergiu Pojoga  пише:
>>
>>> I might be wrong, but isn't ACK part of a transaction? If so, you'd need
>>> to use *setflag*/*isflagset*
>>>
>>> Cheers.
>>>
>>> On Fri, May 24, 2019 at 10:03 AM Володимир Іванець <
>>> volodyaivan...@gmail.com> wrote:
>>>
 Hello everyone!

 I'm not sure if I correctly understood the description of functions
 *isbflagset* and *isbflagset* so once again I'm asking for help :)

 Here is what I do:
 1. Asterisk sends INVITE to Kamailio. Kamailio performs lookup of
 available contacts and then few checks in branch_route. I'm adding 
 *setbflag("17",
 $T_branch_idx);* there.
 2. Later in route[NATMANAGE] I added *if (isbflagset("17",
 $T_branch_idx)) {* and additional logic. *isbflagset* results in *true*
 (for same INVITE).
 3. Finally one of UACs responds with 200 OK. Asterisk generates ACK
 packet that gets in route[NATMANAGE]. *isbflagset* results in *false*.

 So my question is: shouldn't *isbflagset* be *true* for ACK too since
 it's the same branch or I interpreted these fenctions incorrectly?

 Thanks a lot!
 ___
 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 setbflag and isbflagset usage

[Alex Balashov]

Branches are a creature of transactions. It follows that “branch flags” are 
transaction-persistent flags at the branch level, rather than scoped to the 
transaction as a whole. 

ACKs constitute a separate transaction.

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

> On May 24, 2019, at 2:45 PM, Sergiu Pojoga  wrote:
> 
> Then.. I don't know, lol
> 
> But the fact that ACKs belong or not to even different transactions 
> (depending on whether positive or negative reply) must be playing a role here.
> 
> Good luck. 
> 
>> On Fri, May 24, 2019, 12:11 PM Володимир Іванець,  
>> wrote:
>> Hello Sergiu,
>> 
>> We allow multiple contacts to register on single account. They can have 
>> different properties.  On step 1 I'm looking for something specific which 
>> can be different for different contacts. If I understood correctly setbflag 
>> and isbflagset can set and check flags for different branches independently.
>> 
>> Thank you!
>> 
>> пт, 24 трав. 2019 о 18:14 Sergiu Pojoga  пише:
>>> I might be wrong, but isn't ACK part of a transaction? If so, you'd need to 
>>> use setflag/isflagset
>>> 
>>> Cheers.
>>> 
 On Fri, May 24, 2019 at 10:03 AM Володимир Іванець 
  wrote:
 Hello everyone!
 
 I'm not sure if I correctly understood the description of functions 
 isbflagset and isbflagset so once again I'm asking for help :)
 
 Here is what I do:
 1. Asterisk sends INVITE to Kamailio. Kamailio performs lookup of 
 available contacts and then few checks in branch_route. I'm adding 
 setbflag("17", $T_branch_idx); there.
 2. Later in route[NATMANAGE] I added if (isbflagset("17", $T_branch_idx)) 
 { and additional logic. isbflagset results in true (for same INVITE).
 3. Finally one of UACs responds with 200 OK. Asterisk generates ACK packet 
 that gets in route[NATMANAGE]. isbflagset results in false.
 
 So my question is: shouldn't isbflagset be true for ACK too since it's the 
 same branch or I interpreted these fenctions incorrectly?
 
 Thanks a lot!
 ___
 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 setbflag and isbflagset usage

[Sergiu Pojoga]

Then.. I don't know, lol

But the fact that ACKs belong or not to even different transactions
(depending on whether positive or negative reply) must be playing a role
here.

Good luck.

On Fri, May 24, 2019, 12:11 PM Володимир Іванець, 
wrote:

> Hello Sergiu,
>
> We allow multiple contacts to register on single account. They can have
> different properties.  On step 1 I'm looking for something specific which
> can be different for different contacts. If I understood correctly
> *setbflag* and *isbflagset* can set and check flags for different
> branches independently.
>
> Thank you!
>
> пт, 24 трав. 2019 о 18:14 Sergiu Pojoga  пише:
>
>> I might be wrong, but isn't ACK part of a transaction? If so, you'd need
>> to use *setflag*/*isflagset*
>>
>> Cheers.
>>
>> On Fri, May 24, 2019 at 10:03 AM Володимир Іванець <
>> volodyaivan...@gmail.com> wrote:
>>
>>> Hello everyone!
>>>
>>> I'm not sure if I correctly understood the description of functions
>>> *isbflagset* and *isbflagset* so once again I'm asking for help :)
>>>
>>> Here is what I do:
>>> 1. Asterisk sends INVITE to Kamailio. Kamailio performs lookup of
>>> available contacts and then few checks in branch_route. I'm adding 
>>> *setbflag("17",
>>> $T_branch_idx);* there.
>>> 2. Later in route[NATMANAGE] I added *if (isbflagset("17",
>>> $T_branch_idx)) {* and additional logic. *isbflagset* results in *true*
>>> (for same INVITE).
>>> 3. Finally one of UACs responds with 200 OK. Asterisk generates ACK
>>> packet that gets in route[NATMANAGE]. *isbflagset* results in *false*.
>>>
>>> So my question is: shouldn't *isbflagset* be *true* for ACK too since
>>> it's the same branch or I interpreted these fenctions incorrectly?
>>>
>>> Thanks a lot!
>>> ___
>>> 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 setbflag and isbflagset usage

[Володимир Іванець]

Hello Sergiu,

We allow multiple contacts to register on single account. They can have
different properties.  On step 1 I'm looking for something specific which
can be different for different contacts. If I understood correctly
*setbflag* and *isbflagset* can set and check flags for different branches
independently.

Thank you!

пт, 24 трав. 2019 о 18:14 Sergiu Pojoga  пише:

> I might be wrong, but isn't ACK part of a transaction? If so, you'd need
> to use *setflag*/*isflagset*
>
> Cheers.
>
> On Fri, May 24, 2019 at 10:03 AM Володимир Іванець <
> volodyaivan...@gmail.com> wrote:
>
>> Hello everyone!
>>
>> I'm not sure if I correctly understood the description of functions
>> *isbflagset* and *isbflagset* so once again I'm asking for help :)
>>
>> Here is what I do:
>> 1. Asterisk sends INVITE to Kamailio. Kamailio performs lookup of
>> available contacts and then few checks in branch_route. I'm adding 
>> *setbflag("17",
>> $T_branch_idx);* there.
>> 2. Later in route[NATMANAGE] I added *if (isbflagset("17",
>> $T_branch_idx)) {* and additional logic. *isbflagset* results in *true*
>> (for same INVITE).
>> 3. Finally one of UACs responds with 200 OK. Asterisk generates ACK
>> packet that gets in route[NATMANAGE]. *isbflagset* results in *false*.
>>
>> So my question is: shouldn't *isbflagset* be *true* for ACK too since
>> it's the same branch or I interpreted these fenctions incorrectly?
>>
>> Thanks a lot!
>> ___
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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
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Re: [SR-Users] Question about setbflag and isbflagset usage

[Sergiu Pojoga]

I might be wrong, but isn't ACK part of a transaction? If so, you'd need to
use *setflag*/*isflagset*

Cheers.

On Fri, May 24, 2019 at 10:03 AM Володимир Іванець 
wrote:

> Hello everyone!
>
> I'm not sure if I correctly understood the description of functions
> *isbflagset* and *isbflagset* so once again I'm asking for help :)
>
> Here is what I do:
> 1. Asterisk sends INVITE to Kamailio. Kamailio performs lookup of
> available contacts and then few checks in branch_route. I'm adding 
> *setbflag("17",
> $T_branch_idx);* there.
> 2. Later in route[NATMANAGE] I added *if (isbflagset("17",
> $T_branch_idx)) {* and additional logic. *isbflagset* results in *true*
> (for same INVITE).
> 3. Finally one of UACs responds with 200 OK. Asterisk generates ACK packet
> that gets in route[NATMANAGE]. *isbflagset* results in *false*.
>
> So my question is: shouldn't *isbflagset* be *true* for ACK too since
> it's the same branch or I interpreted these fenctions incorrectly?
>
> Thanks a lot!
> ___
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Re: [SR-Users] Question regarding retcode

[Nicolas Breuer]

Try to remove the first Xlog because Xlog returns also a $retcode.



De : sr-users  de la part de Andrew Chen 

Envoyé : mardi 9 avril 2019 23:03:08
À : Kamailio (SER) - Users Mailing List
Objet : [SR-Users] Question regarding retcode

Hey all,

Question regarding the type of retcode.

I ran a curl command and $retcode returned 200.  I check against that variable 
if it's 200 or not.  I tried to check 200 as a string and integer but never 
matches so I ran these lines of code:


xlog("L_INFO", "[$ci][$mi][CSeq $cs] Validating meeting ($rU).  Result: 
$var(result)  Return code: $retcode \n");


if(is_int($retcode))

{

xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is an integer \n");

}


if(typeof("$retcode", "str"))

{

xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is a string \n");

}


None of these xlog outputed.


How do I check for $retcode in if or switch statement?


Thanks

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



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Re: [SR-Users] Question regarding retcode

[Andrew Chen]

Sorry ignore this.  Turns out I had the following sequence of events:

- http_query
- xlog my result
- check $retcode

turns out the $retcode was for the xlog,

My bad.

On Tue, Apr 9, 2019 at 5:03 PM Andrew Chen  wrote:

> Hey all,
>
> Question regarding the type of retcode.
>
> I ran a curl command and $retcode returned 200.  I check against that
> variable if it's 200 or not.  I tried to check 200 as a string and integer
> but never matches so I ran these lines of code:
>
> xlog("L_INFO", "[$ci][$mi][CSeq $cs] Validating meeting ($rU).  
> Result:
> $var(result)  Return code: $retcode \n");
>
>
> if(is_int($retcode))
>
> {
>
> xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is an
> integer \n");
>
> }
>
>
> if(typeof("$retcode", "str"))
>
> {
>
> xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is a string
> \n");
>
> }
>
>
> None of these xlog outputed.
>
>
> How do I check for $retcode in if or switch statement?
>
>
> Thanks
>
> --
> Andy Chen
> Sr. Telephony Lead Engineer
> achen@ fuze.com
>
>
>

-- 
Andy Chen
Sr. Telephony Lead Engineer
415 516 5535 (M)
achen@ fuze.com

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Re: [SR-Users] question about replace_body_all

[Daniel-Constantin Mierla]

Hello,

On 01.12.18 18:27, Wilkins, Steve wrote:
>
> Hello,
>
>  
>
> I am attempting to change the Message Body of an SIP message using
> replace_body_all;  I want to replace all attributes that start with  rtcp.
>
> Here is my code =>
>
>  
>
> *replace_body_all(“rtcp:{1}[0-9]{5,}”, newstring), *and I call
> msg_apply_changes() after this call. Note that I verified that the
> regex is working using *search.*
>
>  
>
> *The problem* is that nothing gets changed. However, what I do get is
> a new string at the bottom of the Message Body with my *newstring*
> concatenated three times, which happens to be the number of times that
> rtcp occurs in the Message Body.
>
>  
>
> Note, I am doing this so that I can add the IP address to the rtcp
> attribute because this is required by a Provider.
>
>  
>
> Any ideas?
>
can you provide a sample INVITE message that exposes the issue? I would
like to try to reproduce and fix if there is any issue in the function,
but it happens that none of the phones I have around send such a= lines...

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference -- www.kamailioworld.com
Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

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Re: [SR-Users] question about replace_body_all

[John Petrini]

I've never used replace_body_all but I've done something similar using
subt_body. One of the issues I've encountered is that you need to
explicitly match line feed characters \n. Also in your replacement you
need to include a carriage return and a new line at the end \r\n or
they'll end up on one line like you're seeing.

replace_body_all may behave differently so this advice might not apply.

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Re: [SR-Users] Question on the new Kamailio v5.x Releases

[Fred Posner]

On 6/27/18 2:34 PM, Ginhoux, Patrick wrote:

Hi,

I always read your email regarding new stable releases.

I am currently working on a project upgrade of multiples servers to the 
most recent OS (Windows, RedHat) and middleware.


In this project, I have upgraded SIP Proxies server running on old 
version of Kamailio, to the 5.0.1 release (over RedHat 7.4).


Effort to make that upgrade and tests have been done with success in the 
last months.


So considering to upgrade to the 5.0.7 or 5.1.4 would be a subsequent 
work I need to evaluate.


I could read all the release notes, but it is also a big challenge to 
determine if fixes are interesting for me.


Based on your experience, is there particular cases, fixed in the new 
release I need to pay attention?


Any comments are welcome.

Regards

Patrick GINHOUX



Hi Patrick,

If you're running 5.0.x then an upgrade to 5.0.7 shouldn't be 
significant (should be very straight forward) and will contain at least 
one major security fix since 5.0.1.


If you installed from git, you can just follow these instructions:

https://kamailio.org/docs/tutorials/5.0.x/kamailio-install-guide-git/#maintenance

If you want to migrate from 5.0.x to 5.1.x, there's a few extra commands 
to do (database changes for example if applicable):


https://www.kamailio.org/wiki/install/upgrade/5.0.x-to-5.1.0

--fred

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Re: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX

[Floimair Florian]

Yes it should, but only if “rewrite_contact” is set to “no” for the endpoint 
(chan_pjsip).




With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

From: sr-users  on behalf of "Wilkins, 
Steve" 
Reply-To: "Kamailio (SER) - Users Mailing List" 
Date: Tuesday, 12. June 2018 at 16:27
To: "Kamailio (SER) - Users Mailing List" 
Subject: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX

Hi All,

Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is 
sending the ACK back but the AOR can’t be found in Kamailio

200 OK Kamailio to Asterisk with a contact of
Contact: 


Asterisk ACK to Kamailio
Request-Line: ACK 
sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob
 SIP/2.0

Shouldn’t Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio 
send Incorrect Contact?
Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR.

Thank you,
-Steve
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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

[Wilkins, Steve]

Hi Alex,

On a WebRTC to WebRTC call (both UAC's registered in Kamailio) =>

- When Asterisk sends the INVITE back to Kamailio, it has the id of the call 
and the IP Address of the laptop the call was made from 
INVITE 
sip:vn768thu@3stum5dgqf9j.invalid;transport=ws;alias=133.148.203.97~53778~6;alias=133.148.203.97~53778~6;ob
 SIP/2.0

- I have been using IP Addresses instead of Domains, as I have had much more 
luck doing it that way.

Should I switch to FQDN and Domains?  If so, may I ask what all would need to 
be changed?

Thank you for your time. 

-Steve

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Alex 
Balashov
Sent: Sunday, May 13, 2018 10:32 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of 
an "INVITE"

That seems correct and sounds like it should work. What is the request URI of 
the INVITE that comes back into Kamailio on the B leg? Does its domain match 
the AOR domain? Do you have your registrar/usrloc set to use domains on 
lookup()?


-- Alex

--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

[Alex Balashov]

That seems correct and sounds like it should work. What is the request
URI of the INVITE that comes back into Kamailio on the B leg? Does its
domain match the AOR domain? Do you have your registrar/usrloc set to
use domains on lookup()?

-- Alex

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

[Wilkins, Steve]

Yes, that is what has me a little boggled.  If I don't, calls do not go through 
so I am obviously confused, but that is how I was finally able to get full 
duplex calls working.

In Asterisk, I have a Kamailio endpoint listening for traffic from Kamailio. 
How would Asterisk know the Contact information if Asterisk does not re-contact 
Kamailio for that Information (since all the registration information is on 
Kamailio).
 
I was under the impression that it is best to keep the registrations on 
Kamailio, why forward them.  Below is my pjsip.conf.
When a user calls 30001@192.21.1.5 (Kamailio IP), the call is registered in 
Kamailio, and Asterisk can contact Kamailio to get the 
Contact information it needs.  


Pjsip.conf

[kamailio](!)
type=endpoint
context=from-internal
transport=transport-tcp
media_address=100.20.30.125 //Asterisk PBX IP
...
aors=kamailio

[kamailio](kamailio)
aors=kamailio

[kamailio]
type=aor
contact=sip:192.21.1.5:5060

[kamailio]
type=identify  ; Must be of type identify (default: "")
endpoint=kamailio
match=192.21.1.5

[30001](webrtc) //tls endpoint
auth=auth30001
aors=30001

[auth30001](auth-userpass)
password=12345
username=30001

[30001](aor-single-reg)
contact=sip:30001@192.21.1.5:5060

Am I totally wrong, even though it works?

Thank you all very much!
-Steve




-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Alex 
Balashov
Sent: Sunday, May 13, 2018 9:19 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of 
an "INVITE"

On Sun, May 13, 2018 at 01:14:08PM +, Wilkins, Steve wrote:

> In my configuration I have to use Asterisk as my PBX, and I use 
> Kamailio in front of Asterisk accepting and inspecting calls.  I have 
> many AORs, for which a phone can register to.  I have noticed that, 
> depending on the call, when a call arrives it gets REGISTERED in 
> Kamailio (I do not forward the registration to Asterisk); Kamailio 
> sends the INVITE to Asterisk, and then Asterisk send the INVITE back 
> to Kamailio.  The calls seem to work fine (Duplex Audi/Video).  I 
> think I know why the INVITE is forwarded to Kamailio but have not been 
> able to work around it; the AOR's all have a Contact of Kamailio and I 
> think this is the reason for the forwarding of the INVITE.  If I don't 
> set Kamailio as the Contact for the AOR's, calls do not work.

Are you sure that you are using the word AOR correctly? 

If you're not forwarding registrations to Asterisk but instead storing them in 
Kamailio's registrar, setting the Contact binding of an AOR to have the address 
of that very same Kamailio registrar would make no sense.

--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

[Alex Balashov]

On Sun, May 13, 2018 at 01:14:08PM +, Wilkins, Steve wrote:

> In my configuration I have to use Asterisk as my PBX, and I use
> Kamailio in front of Asterisk accepting and inspecting calls.  I have
> many AORs, for which a phone can register to.  I have noticed that,
> depending on the call, when a call arrives it gets REGISTERED in
> Kamailio (I do not forward the registration to Asterisk); Kamailio
> sends the INVITE to Asterisk, and then Asterisk send the INVITE back
> to Kamailio.  The calls seem to work fine (Duplex Audi/Video).  I
> think I know why the INVITE is forwarded to Kamailio but have not been
> able to work around it; the AOR's all have a Contact of Kamailio and I
> think this is the reason for the forwarding of the INVITE.  If I don't
> set Kamailio as the Contact for the AOR's, calls do not work.

Are you sure that you are using the word AOR correctly? 

If you're not forwarding registrations to Asterisk but instead storing
them in Kamailio's registrar, setting the Contact binding of an AOR to
have the address of that very same Kamailio registrar would make no
sense.

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

[Wilkins, Steve]

Hello Pan,

Thank you for responding!,

In my configuration I have to use Asterisk as my PBX, and I use Kamailio in 
front of Asterisk accepting and inspecting calls.
I have many AORs, for which a phone can register to.  I have noticed that, 
depending on the call, when a call arrives it gets REGISTERED in
Kamailio (I do not forward the registration to Asterisk); Kamailio sends the 
INVITE to Asterisk, and then Asterisk send the INVITE back to
Kamailio.  The calls seem to work fine (Duplex Audi/Video).  I think I know why 
the INVITE is forwarded to Kamailio but have not been able
to work around it; the AOR's all have a Contact of Kamailio and I think this is 
the reason for the forwarding of the INVITE.  If I don't set Kamailio as the 
Contact for the AOR's, calls do not work.

Thanks Again,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, May 11, 2018 10:20 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of 
an "INVITE"

Hello Steve.

What are you trying to achieve?

The call could go from client A to Kamailio to client B. No need to involve 
Asterisk. If you need PBX functionality, the INVITE needs to be routed to 
Asterisk, which will most likely answer the call and then set up a new call to 
client B. As Asterisk doesn't know where client B is, it needs to route this 
new call to Kamailio where client B is registered. It's possible for Asterisk 
to know where client B is but that solves nothing and may create other problems.

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: onsdag 9. mai 2018 19:15
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Question about using Kamailio and Asterisk and flow of an 
"INVITE"

Hello All,

I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using 
Kamailio and Asterisk.

Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 
from WebRTC Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the 
Invite back to Kamailio.  Also depending on
The version of Asterisk, the INVITE will then get forwarded to the AOR that is 
registered in Kamailio for the called number.
Does this seem correct?  It seems like there is an extra hop in there.

The reason I am now very curious now is because everything works fine if using 
Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3
I get the extra hop and call is dropped after 30 seconds.

I would appreciate any thoughts on this.

Thank you in advance.



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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

[Pan Christensen]

Hello Steve.

What are you trying to achieve?

The call could go from client A to Kamailio to client B. No need to involve 
Asterisk. If you need PBX functionality, the INVITE needs to be routed to 
Asterisk, which will most likely answer the call and then set up a new call to 
client B. As Asterisk doesn't know where client B is, it needs to route this 
new call to Kamailio where client B is registered. It's possible for Asterisk 
to know where client B is but that solves nothing and may create other problems.

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users  On Behalf Of Wilkins, Steve
Sent: onsdag 9. mai 2018 19:15
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Question about using Kamailio and Asterisk and flow of an 
"INVITE"

Hello All,

I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using 
Kamailio and Asterisk.

Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 
from WebRTC Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the 
Invite back to Kamailio.  Also depending on
The version of Asterisk, the INVITE will then get forwarded to the AOR that is 
registered in Kamailio for the called number.
Does this seem correct?  It seems like there is an extra hop in there.

The reason I am now very curious now is because everything works fine if using 
Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3
I get the extra hop and call is dropped after 30 seconds.

I would appreciate any thoughts on this.

Thank you in advance.



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Re: [SR-Users] question on TM transaction matching

[Kelvin Chua]

yes i agree quite difficult to trace, here is the invite of the non-working
one

U 2018/04/19 04:39:42.316735 w.x.y.z:44308 -> a.b.c.d:6060
INVITE sip:408@a.b.c.d:6060;transport=UDP SIP/2.0.
Via: SIP/2.0/UDP w.x.y.z:44308;branch=z9hG4bK-524287-1---bac673a0af203e10;
rport.
Max-Forwards: 70.
Contact: .
To: .
From: ;tag=88ab0030.
Call-ID: qXQ2PEtXgehz4CQYo2ryLg...
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE.
Content-Type: application/sdp.
User-Agent: Zoiper rv2.8.87-mod.
Allow-Events: presence, kpml, talk.
Content-Length: 243.
.
v=0.
o=Zoiper 0 0 IN IP4 192.168.1.234.
s=Zoiper.
c=IN IP4 192.168.1.234.
t=0 0.
m=audio 38678 RTP/AVP 3 101 0 8.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=sendrecv.


U 2018/04/19 04:39:42.318569 a.b.c.d:6060 -> w.x.y.z:44308
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP w.x.y.z:44308;branch=z9hG4bK-524287-1---bac673a0af203e10;
rport=44308;received=w.x.y.z.
To: .
From: ;tag=88ab0030.
Call-ID: qXQ2PEtXgehz4CQYo2ryLg...
CSeq: 1 INVITE.




Kelvin Chua

On Thu, Apr 19, 2018 at 5:37 PM, Daniel-Constantin Mierla  wrote:

> Send the full trace for not working scenario, from invite to cancel, it is
> hard to track from two partial traces, small bits can make difference here,
> so if these call-id, branch, tags are different, then I have to work on
> guessing, ...
>
> Cheers,
> Daniel
>
> On 19.04.18 11:00, Kelvin Chua wrote:
>
> yes i have it, sorry i missed that part, they are identical except for
> Call-ID, tag, branch
>
> U 2018/04/19 05:31:30.319735 w.x.y.z:34439 -> a.b.c.d:6060
> INVITE sip:408@a.b.c.d:6060;transport=UDP SIP/2.0.
> Via: SIP/2.0/UDP w.x.y.z:34439;branch=z9hG4bK-524287-1---b186786334147b7d;
> rport.
> Max-Forwards: 70.
> Contact: .
> To: .
> From: ;tag=d75e236e.
> Call-ID: 7O090_g-Ndh_I22Zi5ZOhQ...
> CSeq: 1 INVITE.
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE.
> Content-Type: application/sdp.
> User-Agent: Zoiper rv2.8.87-mod.
> Allow-Events: presence, kpml, talk.
> Content-Length: 243.
> .
> v=0.
> o=Zoiper 0 0 IN IP4 192.168.1.234.
> s=Zoiper.
> c=IN IP4 192.168.1.234.
> t=0 0.
> m=audio 38678 RTP/AVP 3 101 0 8.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=sendrecv.
>
>
> U 2018/04/19 05:31:30.320283 a.b.c.d:6060 -> w.x.y.z:34439
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP w.x.y.z:34439;branch=z9hG4bK-524287-1---b186786334147b7d;
> rport=34439;received=w.x.y.z.
> To: .
> From: ;tag=d75e236e.
> Call-ID: 7O090_g-Ndh_I22Zi5ZOhQ...
> CSeq: 1 INVITE.
> Server: kamailio (5.1.2 (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> Kelvin Chua
>
> On Thu, Apr 19, 2018 at 4:14 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> The INVITE from w.x.y.z:44308 -> a.b.c.d:6060 is not given. That's the
>> one that is needed in order to match the CANCEL, which comes from w.x.y.z.
>>
>> Can you provide it?
>> Cheers,
>> Daniel
>>
>>
>> On 19.04.18 09:47, Kelvin Chua wrote:
>>
>> *here is an ngrep of the working one*
>>
>> U 2018/04/19 05:31:31.112392 a.b.c.d:6060 -> a.b.c.d:5090
>> INVITE sip:408@a.b.c.d:6060;transport=UDP SIP/2.0.
>> Record-Route: .
>> Via: SIP/2.0/UDP a.b.c.d:6060;branch=z9hG4bK638
>> a.ebce2b6669d2c54149bcc613c89d6510.1.
>> Via: SIP/2.0/UDP w.x.y.z:34439;received=w.x.y.z
>> ;branch=z9hG4bK-524287-1---4c57b6fa8ff74676;rport=34439.
>> Max-Forwards: 70.
>> Contact: .
>> To: .
>> From: ;tag=d75e236e.
>> Call-ID: 7O090_g-Ndh_I22Zi5ZOhQ...
>> P-Called-Identity: 408.
>> CSeq: 2 INVITE.
>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
>> SUBSCRIBE.
>> Content-Type: application/sdp.
>> Proxy-Authorization: Digest username="kelvin",realm="domain.com
>> ",nonce="3861e6cdaf8a44cce98721e3c8dda428",uri="sip:
>> 408@a.b.c.d:6060;transport=UDP",response="b7a365bd3565073
>> 6c45d2048d924ea0b",cnonce="492da67adf4a5853f0aa012c2fa5a757"
>> ,nc=0001,qop=auth,algorithm=MD5.
>> User-Agent: Zoiper rv2.8.87-mod.
>> Allow-Events: presence, kpml, talk.
>> Content-Length: 255.
>> P-Called-Party-ID: .
>> .
>> v=0.
>> o=Zoiper 0 0 IN IP4 a.b.c.d.
>> s=Zoiper.
>> c=IN IP4 a.b.c.d.
>> t=0 0.
>> m=audio 19592 RTP/AVP 3 101 0 8.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:8 PCMA/8000.
>> a=sendrecv.
>> a=rtcp:19593.
>>
>>
>> U 2018/04/19 05:31:31.180853 a.b.c.d:5090 -> a.b.c.d:6060
>> SIP/2.0 100 Trying.
>> Via: SIP/2.0/UDP a.b.c.d:6060;branch=z9hG4bK638
>> a.ebce2b6669d2c54149bcc613c89d6510.1.
>> Via: SIP/2.0/UDP w.x.y.z:34439;received=w.x.y.z
>> ;branch=z9hG4bK-524287-1---4c57b6fa8ff74676;rport=34439.
>> To: .
>> From: ;tag=d75e236e.
>> Call-ID: 7O090_g-Ndh_I22Zi5ZOhQ...
>> CSeq: 2 INVITE.
>> Server: kamailio (5.1.2 (x86_64/linux)).
>> Content-Length: 0.
>> .
>>
>>
>> U 2018/04/19 05:31:35.134023 a.b.c.d:5090 -> a.b.c.d:6060
>> SIP/2.0 183 Sessi