[SR-Users] CallControl and MediaProxy

Reda Aouad reda.aouad at gmail.com
Mon Feb 27 18:39:27 CET 2012


Ok thanks Daniel.

I'll do what you suggested and we'll see how to proceed.

Thanks again
Reda

On 27 févr. 2012, at 17:46, Daniel-Constantin Mierla <miconda at gmail.com>
wrote:

 Hello,

On 2/26/12 11:01 AM, Reda Aouad wrote:

Daniel, I sent you the log file by private email, for privacy concerns.

ok, no time yet to look over them, but as you say it does not call
mediaproxy explicitely from config and callcontrol, maybe you should look
inside mediaproxy module, might be automatically executed on dialog or tm
module callbacks -- then the patch has to be done for mediaproxy module.

 I took a quick look at the log files, and didn't see any call to
mediaproxy module.

 Though, the mediaproxy log file indicates a media session engaged, and
CDRTool shows media information for the established session.

 The mediaproxy module is loaded, but no function from the module is called
at all.

It can be via some internal callbacks registered to some modules such as tm
and dialog, or even script execution callbacks... so look over the code of
mediaproxy module. You can run kamailio with debug=3 and from the logs you
should get some hints about what is the way media proxy is executed.

Cheers,
Daniel


 Thanks
 Reda



On Sat, Feb 25, 2012 at 18:58, Reda Aouad <reda.aouad at gmail.com> wrote:

> I looked in detail in the source of the call control module and didn't
> find anything related to the mediaproxy module neither..
> I'll send you the output of cfgtrace soon.
>
>  Reda
>
>
>
> On Fri, Feb 24, 2012 at 23:46, Daniel-Constantin Mierla <miconda at gmail.com
> > wrote:
>
>>  A quick grep over the callcontrol sources didn't reveal a point where
>> mediaproxy is engaged.
>>
>> Can you load debugger module and enable cfgtrace, then run such a
>> scenario and send out the output from cfgtrace to see all the config
>> actions executed?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 2/24/12 7:56 PM, Reda Aouad wrote:
>>
>> It would be great Daniel if you could do it.
>> I will be more than happy to test it.
>> A function parameter would be more flexible than a module parameter.
>> I expect it shouldn't affect the behavior of the external application
>> CallControl.
>>
>>  Thanks in advance :)
>>
>>  Reda
>>
>>
>>
>> On Fri, Feb 24, 2012 at 08:09, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> I am not using mediaproxy at all (but nathelper/rtpproxy), neither the
>>> call control module, but making an option (module parameter or function
>>> parameter) for call control to bind to another module like media proxy,
>>> should not be big deal if it is all you are looking for -- I can look over
>>> it and send a patch if you are going to help testing it. I cannot do it
>>> these days, though, being out of the office.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 2/23/12 8:59 PM, Reda Aouad wrote:
>>>
>>>  First, I am posting about the wrong behavior of CallControl (or most
>>> probably Kamailio modules) which leaves no option. I should be the only
>>> one deciding about how to handle timeouts. If I decide to take some risk,
>>> no module should oblige me to do otherwise.
>>>
>>>  Mediaproxy detects ONLY RTP timeouts from BOTH parties, because linux
>>> conntrack rules it uses are bi-directional. If a single party stops sending
>>> RTP for whatever reason (connection lost, codec with silence detection
>>> used, ....), mediaproxy doesn't care and doesn't act upon it. This is a
>>> feature, and a wanted one, to mainly support voice-detecting codecs. Think
>>> also about conferences for example, in which only a single person talks for
>>> a long time while others are silent and don't send RTP.
>>>
>>>  Single-side RTP timeout because of a real problem (loosing network
>>> connection for example) should be handled with other methods, such as SIP
>>> session timers.
>>>
>>>  MY POINT IS : I don't see it practical to handle RTP flows for EVERY
>>> call to handle the least probable scenario: an RTP timeout from both (or
>>> all) parties.
>>>
>>>  If I understood well, mediaproxy updates the CDR when it detects an
>>> RTP timeout from both parties. CallControl can look in the CDR to debit the
>>> correct balance, instead of attaching itself to the dialog module to detect
>>> dialog termination.
>>>
>>>  This is an extract from the call_control module :
>>>
>>>  Even when mediaproxy is unable to end the dialog because it was not
>>> started with engage_media_proxy(), the callcantrol application is still
>>> able to detect calls that did timeout sending media, by looking in the
>>> radius accounting records for entries recorded by mediaproxy for calls that
>>> did timeout. These calls will also be ended gracefully by the callcontrol
>>> application itself.
>>>
>>>
>>>  Unless there is something I miss..
>>>
>>>  I also opened a bug about the issue because call_control doesn't have
>>> the same behavior with OpenSips. It doesn't force mediaproxy.
>>>
>>>  Reda
>>>
>>>
>>>
>>> On Thu, Feb 23, 2012 at 20:00, Jeff Brower <jbrower at signalogic.com>wrote:
>>>
>>>> Reda-
>>>>
>>>> > It's clear but not necessary. It can look at radius records fixed by
>>>> > mediaproxy on RTP timeout to debit the correct balance as well. And
>>>> why
>>>> > also force it on postpaid calls which it doesn't control at all ?
>>>>
>>>>  I don't understand how you plan to tear down Kamailio calls that
>>>> suffer RTP time-out?
>>>>
>>>> -Jeff
>>>>
>>>> > What happens is cost and performance issues for additional calls
>>>> passing
>>>> > through my mediaproxy server, which I didn't plan for at first. No
>>>> audio
>>>> > issue at all.
>>>> >
>>>> > Reda
>>>> >
>>>> >
>>>> >
>>>> > On Thu, Feb 23, 2012 at 11:58, Sammy Govind <govoiper at gmail.com>
>>>> wrote:
>>>> >
>>>> >> Reading from the module docs its clear why it needs to engage
>>>> media/rtp
>>>> >> proxy to start,stop billing or timer of a call. so what happens when
>>>> it
>>>> >> engages mediaproxy on unwanted calls !? audio-issues?
>>>> >>
>>>> >>
>>>> >> On Thu, Feb 23, 2012 at 1:21 PM, Reda Aouad <reda.aouad at gmail.com>
>>>> wrote:
>>>> >>
>>>> >>> Thanks Sammy. I didn't get any reply yet.
>>>> >>>
>>>> >>> CallControl is an application used with CDRTool for prepaid calls.
>>>> It
>>>> >>> calculates the maximum call duration based on the user's balance.
>>>> Once the
>>>> >>> call's duration limit is reached, it sends BYE to both calling
>>>> parties,
>>>> >>> terminating the call. At the end of a prepaid call, terminated
>>>> either by
>>>> >>> the user or by CallControl, it debits the user's balance according
>>>> to the
>>>> >>> call's duration.
>>>> >>>
>>>> >>> The Call_Control module interfaces with this external application.
>>>> >>>
>>>> >>> call_control function is called in Kamailio's cfg to check if the
>>>> user
>>>> >>> has prepaid or postpaid account, and get the max call duration for
>>>> prepaid
>>>> >>> users. CallControl controls only prepaid calls, not postpaid ones.
>>>> >>>
>>>> >>> So call control and NAT traversal using mediaproxy are two
>>>> differents
>>>> >>> things which i can't link, since I don't want mediaproxy for every
>>>> call.
>>>> >>> And since the function call_control is called on every invite before
>>>> >>> knowing if the user has a prepaid account or not, it engages
>>>> mediaproxy for
>>>> >>> every call.
>>>> >>>
>>>> >>> CallControl relies on mediaproxy to detect RTP timeouts and debit
>>>> the
>>>> >>> correct balance from a prepaid account based on the last instant the
>>>> >>> mediaproxy saw an RTP packet.
>>>> >>>
>>>> >>> But why to force using mediaproxy with no choice? And why to force
>>>> it for
>>>> >>> every call, whether it falls under CallControl's control or not?
>>>> >>>
>>>> >>> I am using Kamailio 3.2.
>>>> >>>
>>>> >>>
>>>> >>> Reda
>>>> >>>
>>>> >>> On 23 févr. 2012, at 07:21, Sammy Govind <govoiper at gmail.com>
>>>> wrote:
>>>> >>>
>>>> >>> Hi,
>>>> >>> I can see you posting multiple times on both proxies listings so
>>>> I'm sure
>>>> >>> you havent heard back from anyone.I am not at all familiar with your
>>>> >>> functions in email but could it be possible for you to determine on
>>>> which
>>>> >>> calls you need to engage mediaproxy and on which not to, then on
>>>> the base
>>>> >>> of that flag use the call_control function !
>>>> >>> your problem is complicated for me atleast. I hope somebody could
>>>> answer
>>>> >>> you accurately and precisely.
>>>> >>>
>>>> >>> btw, what are you using in real? opensips or kamailio, which
>>>> version? and
>>>> >>> in what context you need to use the call_control function?
>>>> >>>
>>>> >>> Thanks,
>>>> >>> Sammy
>>>> >>>
>>>> >>> On Thu, Feb 23, 2012 at 12:45 AM, Reda Aouad <reda.aouad at gmail.com
>>>> >wrote:
>>>> >>>
>>>> >>>> Hi,
>>>> >>>>
>>>> >>>> When I use the function call_control( ) of the call_control
>>>> module, it
>>>> >>>> automatically engages mediaproxy if it finds the mediaproxy module
>>>> loaded.
>>>> >>>> If the mediaproxy module is not loaded, call_control doesn't even
>>>> try to
>>>> >>>> engage it.
>>>> >>>>
>>>> >>>> I need mediaproxy for NAT traversal in some cases, but don't want
>>>> it to
>>>> >>>> be engaged on every call.
>>>> >>>>
>>>> >>>> How can I disable this behavior?
>>>> >>>>
>>>> >>>> Thanks
>>>> >>>> Reda
>>>> >>>>
>>>> >>>> _______________________________________________
>>>> >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>> list
>>>> >>>> sr-users at lists.sip-router.org
>>>> >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>> >>>>
>>>> >>>>
>>>> >>> _______________________________________________
>>>> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>> list
>>>> >>> sr-users at lists.sip-router.org
>>>> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>> >>>
>>>> >>>
>>>> >>> _______________________________________________
>>>> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>> list
>>>> >>> sr-users at lists.sip-router.org
>>>> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>> >>>
>>>> >>>
>>>> >>
>>>> >> _______________________________________________
>>>> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>> list
>>>> >> sr-users at lists.sip-router.org
>>>> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>> >>
>>>> >>
>>>> > _______________________________________________
>>>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>> list
>>>> > sr-users at lists.sip-router.org
>>>> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>> >
>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>   --
>>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>>>
>>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>


_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


-- 
Daniel-Constantin Mierla --
http://www.asipto.comhttp://linkedin.com/in/miconda --
http://twitter.com/miconda
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20120227/b168d086/attachment-0001.htm>


More information about the sr-users mailing list