22 Feb
2012
22 Feb
'12
8:45 p.m.
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
23 Feb
23 Feb
7:19 a.m.
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(a)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
_______________________________________________
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9:21 a.m.
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(a)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(a)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
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11:58 a.m.
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(a)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(a)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(a)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
_______________________________________________
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sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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_______________________________________________
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12:03 p.m.
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 ?
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(a)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(a)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(a)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(a)gmail.com>wrote;wrote:
_______________________________________________
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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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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8 p.m.
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(a)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(a)gmail.com> wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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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(a)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(a)gmail.com>wrote;wrote:
_______________________________________________
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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(a)lists.sip-router.org
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_______________________________________________
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_______________________________________________
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8:59 p.m.
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(a)signalogic.com> wrote:
Reda-
loaded.
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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(a)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(a)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(a)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(a)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 >>> 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(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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24 Feb
24 Feb
8:09 a.m.
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(a)signalogic.com
<mailto:jbrower@signalogic.com>> wrote:
Reda-
--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
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(a)gmail.com
<mailto:govoiper@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(a)gmail.com
<mailto:reda.aouad@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(a)gmail.com
<mailto:govoiper@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(a)gmail.com
<mailto:reda.aouad@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(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
>>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
list
>> sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
list
>> sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list
> sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
_______________________________________________
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<mailto:sr-users@lists.sip-router.org>
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_______________________________________________
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7:56 p.m.
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(a)gmail.com>wrote;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(a)signalogic.com> wrote:
Reda-
it
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda --
http://twitter.com/miconda
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(a)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(a)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(a)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(a)gmail.com
wrote:
>>
>>> Hi,
>>>
>>> When I use the function call_control( ) of the call_control module, >>> 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(a)lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
>> sr-users(a)lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
>
sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
_______________________________________________
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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11:46 p.m.
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(a)gmail.com <mailto:miconda@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:
--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
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(a)signalogic.com <mailto:jbrower@signalogic.com>> wrote:
Reda-
--
Daniel-Constantin Mierla --http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 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(a)gmail.com
<mailto:govoiper@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(a)gmail.com
<mailto:reda.aouad@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(a)gmail.com
<mailto:govoiper@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(a)gmail.com <mailto:reda.aouad@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(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
>>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>
_______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing list
>> sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing
list
>> sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
list
> sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
_______________________________________________
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mailing list
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<mailto:sr-users@lists.sip-router.org>
_______________________________________________
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25 Feb
25 Feb
6:58 p.m.
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(a)gmail.com>wrote;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(a)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(a)signalogic.com>wrote;wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda --
http://twitter.com/miconda
Reda-
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda --
http://twitter.com/miconda
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(a)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(a)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(a)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(a)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(a)lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
>> sr-users(a)lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
>> sr-users(a)lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
_______________________________________________
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sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
26 Feb
26 Feb
11:01 a.m.
Daniel, I sent you the log file by private email, for privacy concerns.
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.
Thanks
Reda
On Sat, Feb 25, 2012 at 18:58, Reda Aouad <reda.aouad(a)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(a)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(a)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(a)signalogic.com>wrote;wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda --
http://twitter.com/miconda
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(a)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(a)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(a)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(a)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(a)lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
>>> sr-users(a)lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
>>> sr-users(a)lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
>> sr-users(a)lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
_______________________________________________
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listsr-users@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
27 Feb
27 Feb
5:46 p.m.
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(a)gmail.com
<mailto:reda.aouad@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(a)gmail.com <mailto:miconda@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:
--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
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(a)gmail.com <mailto:miconda@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:
--
Daniel-Constantin Mierla --http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 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(a)signalogic.com <mailto:jbrower@signalogic.com>>
wrote:
Reda-
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla --http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 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(a)gmail.com
<mailto:govoiper@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(a)gmail.com <mailto:reda.aouad@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(a)gmail.com <mailto:govoiper@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(a)gmail.com
<mailto:reda.aouad@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(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
>>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>
_______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER)
- sr-users mailing list
>> sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>>
_______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER)
- sr-users mailing list
>> sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing list
> sr-users(a)lists.sip-router.org
<mailto:sr-users@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
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<mailto:sr-users@lists.sip-router.org>
4449
days inactive
4454
days old
12 comments
4 participants
participants (4)
-
Daniel-Constantin Mierla -
Jeff Brower -
Reda Aouad -
Sammy Govind