[OpenSIPS-Users] rtpproxy n flag throws "Failed to get dialog"

[M S]

Hello again :)
Rtpproxy documentation says:
"for all the calls that require notification, the rtpproxy_engage(),
rtpproxy_offer() and rtpproxy_answer() functions must be called with the “n”
flag"

I'm running rtpproxy_offer("froc") & rtpproxy_response("froc") and opensips
3.1.13, but as soon as I add n flag to both as rtpproxy_offer("frocn") &
rtpproxy_response("frocn"), I see below errors:

ERROR:rtpproxy:force_rtp_proxy_body: Failed to get dialog

And when I remove n flag, the error is gone. A bug maybe?

Thanks!
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Re: [OpenSIPS-Users] RTPPROXY / OPENSIPS

[Bogdan-Andrei Iancu]

Hi Wadii,

I haven;t checked the implementation, but the rtpproxy_engage should 
take care of the 183 with SDP. Have you tested it?


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
  https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/

On 12/23/22 9:42 PM, Wadii ELMAJDI | Evenmedia wrote:

hello , i do have a question related to rtpproxy module documentation.

The doc describes that rewriting sdp body should happen during either 
INVITE , 200 OK or ACK.
In the case of SDP presence on invite <=> 200 , one should 
rtpproxy_offer during the invite and rtpproxy_answer during the 200 OK.


"Documentation : RewritesSDPbody to ensure that media is passed 
through anRTPproxy. To be invoked on INVITE for the cases the SDPs are 
in INVITE and 200 OK and on 200 OK when SDPs are in 200 OK and ACK."


But sometimes opensips receives a 183 Session Progress containing SDP 
before the 200 which i think is related to earlymedia .
I think those sdp packets should also be rewritten with the right rtp 
proxy ip/port. In which case the doc should mention "SDPs are in 
INVITE and 200 OK /183 Session Progress".


My second question : since we can handle most cases with 
rtpproxy_offer/answer methods, what is the purpose of rtpproxy_engage 
? is it to ease the management i described above ?


Thank you



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Re: [OpenSIPS-Users] RTPPROXY / OPENSIPS

[Saint Michael]

I have a lot of calls that should work with RTPPROXY and instead I get dead air.
Maybe this is the issue.

On Fri, Dec 23, 2022 at 2:45 PM Wadii ELMAJDI | Evenmedia
 wrote:
>
> hello , i do have a question related to rtpproxy module documentation.
>
> The doc describes that rewriting sdp body should happen during either INVITE 
> , 200 OK or ACK.
> In the case of SDP presence on invite <=> 200 , one should rtpproxy_offer 
> during the invite and rtpproxy_answer during the 200 OK.
>
> "Documentation : Rewrites SDP body to ensure that media is passed through an 
> RTP proxy. To be invoked on INVITE for the cases the SDPs are in INVITE and 
> 200 OK and on 200 OK when SDPs are in 200 OK and ACK."
>
> But sometimes opensips receives a 183 Session Progress containing SDP before 
> the 200 which i think is related to earlymedia .
> I think those sdp packets should also be rewritten with the right rtp proxy 
> ip/port. In which case the doc should mention "SDPs are in INVITE and 200 OK 
> /183 Session Progress".
>
> My second question : since we can handle most cases with 
> rtpproxy_offer/answer methods, what is the purpose of rtpproxy_engage ? is it 
> to ease the management i described above ?
>
> Thank you
>
>
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[OpenSIPS-Users] RTPPROXY / OPENSIPS

[Wadii ELMAJDI | Evenmedia]

hello , i do have a question related to rtpproxy module documentation.

The doc describes that rewriting sdp body should happen during either INVITE , 
200 OK or ACK.
In the case of SDP presence on invite <=> 200 , one should rtpproxy_offer 
during the invite and rtpproxy_answer during the 200 OK.

"Documentation : Rewrites SDP body to ensure that media is passed through an 
RTP proxy. To be invoked on INVITE for the cases the SDPs are in INVITE and 200 
OK and on 200 OK when SDPs are in 200 OK and ACK."

But sometimes opensips receives a 183 Session Progress containing SDP before 
the 200 which i think is related to earlymedia .
I think those sdp packets should also be rewritten with the right rtp proxy 
ip/port. In which case the doc should mention "SDPs are in INVITE and 200 OK 
/183 Session Progress".

My second question : since we can handle most cases with rtpproxy_offer/answer 
methods, what is the purpose of rtpproxy_engage ? is it to ease the management 
i described above ?

Thank you


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Re: [OpenSIPS-Users] RTPProxy Docker Image

[Maxim Sobolev]

Liviu has done some exploration on getting things handled on Kubernetes.
His great presentation is available here:
https://youtu.be/JwO0UmauuT4?t=13034

-Max

On Wed, Dec 21, 2022, 5:04 PM Terrance Devor  wrote:

> Hello David, Similar to what we have with LXC
>
> OpenSIPS - Proxy, Edge Switch, Managing DIDs and Termination routes, CDR,
> LB to Asterisk
> Asterisk - PBX, IVR
> RTPProxy - Media Relay
>
> Everything containerized using docker and deployed to our k8s cluster.
>
> I would appreciate speaking to anyone that has experience in successfully,
> or failed, in trying to do this
>
> On Wed, Dec 21, 2022 at 7:57 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> Can you explain more? I.e: params and such?
>> Thanks!
>>
>> On Tue, 20 Dec 2022 at 22:29, Saint Michael  wrote:
>>
>>> Opensips+ RTPProxy only works fine with plain LXC containers,
>>> privileged, which basically have access to all the resources of the
>>> box.
>>> That is the model I use with great success.
>>>
>>> On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff 
>>> wrote:
>>> >
>>> > Hello Terrance,
>>> > I wouldn't really recommend this. RTPProxy is going to use a lot of
>>> ports in a very large range. That just doesn't work great in docker, but
>>> even worse in K8S.
>>> >
>>> > I personally would put the RTPProxy outside of K8S. While you might be
>>> able to get it to work, you are likely going against some basic design
>>> concepts in containerization. I feel like the tech should propel the
>>> solution and not be a hindrance to it. In this case, I'm not sure that K8S
>>> is buying you anything of value, but instead creating architectural
>>> challenges.
>>> >
>>> > I'd love to hear feedback or experiences from others. There's always
>>> something to learn :)
>>> > -Brett
>>> >
>>> > On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor 
>>> wrote:
>>> >>
>>> >> Was it something I said?
>>> >>
>>> >> Terrance
>>> >>
>>> >> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor 
>>> wrote:
>>> >>>
>>> >>> Hello Everyone,
>>> >>>
>>> >>> Wow! Blast from the past... I am a long time member of this list,
>>> been a while.
>>> >>>
>>> >>> Question, anyone successful in deploying RTPProxy to a dockerized
>>> environment? Preferably to a Kubernetes managed environment.
>>> >>>
>>> >>> Please Help Team :)
>>> >>>
>>> >>> Kind Regards,
>>> >>> Terrance
>>> >>
>>> >> ___
>>> >> Users mailing list
>>> >> Users@lists.opensips.org
>>> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> >
>>> > ___
>>> > Users mailing list
>>> > Users@lists.opensips.org
>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> ___
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>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>> ___
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Re: [OpenSIPS-Users] RTPProxy Docker Image

[Terrance Devor]

Hello David, Similar to what we have with LXC

OpenSIPS - Proxy, Edge Switch, Managing DIDs and Termination routes, CDR,
LB to Asterisk
Asterisk - PBX, IVR
RTPProxy - Media Relay

Everything containerized using docker and deployed to our k8s cluster.

I would appreciate speaking to anyone that has experience in successfully,
or failed, in trying to do this

On Wed, Dec 21, 2022 at 7:57 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> Can you explain more? I.e: params and such?
> Thanks!
>
> On Tue, 20 Dec 2022 at 22:29, Saint Michael  wrote:
>
>> Opensips+ RTPProxy only works fine with plain LXC containers,
>> privileged, which basically have access to all the resources of the
>> box.
>> That is the model I use with great success.
>>
>> On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff 
>> wrote:
>> >
>> > Hello Terrance,
>> > I wouldn't really recommend this. RTPProxy is going to use a lot of
>> ports in a very large range. That just doesn't work great in docker, but
>> even worse in K8S.
>> >
>> > I personally would put the RTPProxy outside of K8S. While you might be
>> able to get it to work, you are likely going against some basic design
>> concepts in containerization. I feel like the tech should propel the
>> solution and not be a hindrance to it. In this case, I'm not sure that K8S
>> is buying you anything of value, but instead creating architectural
>> challenges.
>> >
>> > I'd love to hear feedback or experiences from others. There's always
>> something to learn :)
>> > -Brett
>> >
>> > On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor 
>> wrote:
>> >>
>> >> Was it something I said?
>> >>
>> >> Terrance
>> >>
>> >> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor 
>> wrote:
>> >>>
>> >>> Hello Everyone,
>> >>>
>> >>> Wow! Blast from the past... I am a long time member of this list,
>> been a while.
>> >>>
>> >>> Question, anyone successful in deploying RTPProxy to a dockerized
>> environment? Preferably to a Kubernetes managed environment.
>> >>>
>> >>> Please Help Team :)
>> >>>
>> >>> Kind Regards,
>> >>> Terrance
>> >>
>> >> ___
>> >> Users mailing list
>> >> Users@lists.opensips.org
>> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>> > ___
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
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Re: [OpenSIPS-Users] RTPProxy Docker Image

[David Villasmil]

Can you explain more? I.e: params and such?
Thanks!

On Tue, 20 Dec 2022 at 22:29, Saint Michael  wrote:

> Opensips+ RTPProxy only works fine with plain LXC containers,
> privileged, which basically have access to all the resources of the
> box.
> That is the model I use with great success.
>
> On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff  wrote:
> >
> > Hello Terrance,
> > I wouldn't really recommend this. RTPProxy is going to use a lot of
> ports in a very large range. That just doesn't work great in docker, but
> even worse in K8S.
> >
> > I personally would put the RTPProxy outside of K8S. While you might be
> able to get it to work, you are likely going against some basic design
> concepts in containerization. I feel like the tech should propel the
> solution and not be a hindrance to it. In this case, I'm not sure that K8S
> is buying you anything of value, but instead creating architectural
> challenges.
> >
> > I'd love to hear feedback or experiences from others. There's always
> something to learn :)
> > -Brett
> >
> > On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor 
> wrote:
> >>
> >> Was it something I said?
> >>
> >> Terrance
> >>
> >> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor 
> wrote:
> >>>
> >>> Hello Everyone,
> >>>
> >>> Wow! Blast from the past... I am a long time member of this list, been
> a while.
> >>>
> >>> Question, anyone successful in deploying RTPProxy to a dockerized
> environment? Preferably to a Kubernetes managed environment.
> >>>
> >>> Please Help Team :)
> >>>
> >>> Kind Regards,
> >>> Terrance
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
-- 
Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
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Re: [OpenSIPS-Users] RTPProxy Docker Image

[Terrance Devor]

Hello Bret, another approach we are thinking about is put RTPProxy on a
VPC. As for Opensips and Asterisk, they can live on a k8s with the
understanding that they will not deal with media directly.

If anyone can share their experience, I would be interested in hearing from
you.

Do the RTP guys still follow this mailing list?

On Tue, Dec 20, 2022 at 4:31 PM Saint Michael  wrote:

> Opensips+ RTPProxy only works fine with plain LXC containers,
> privileged, which basically have access to all the resources of the
> box.
> That is the model I use with great success.
>
> On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff  wrote:
> >
> > Hello Terrance,
> > I wouldn't really recommend this. RTPProxy is going to use a lot of
> ports in a very large range. That just doesn't work great in docker, but
> even worse in K8S.
> >
> > I personally would put the RTPProxy outside of K8S. While you might be
> able to get it to work, you are likely going against some basic design
> concepts in containerization. I feel like the tech should propel the
> solution and not be a hindrance to it. In this case, I'm not sure that K8S
> is buying you anything of value, but instead creating architectural
> challenges.
> >
> > I'd love to hear feedback or experiences from others. There's always
> something to learn :)
> > -Brett
> >
> > On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor 
> wrote:
> >>
> >> Was it something I said?
> >>
> >> Terrance
> >>
> >> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor 
> wrote:
> >>>
> >>> Hello Everyone,
> >>>
> >>> Wow! Blast from the past... I am a long time member of this list, been
> a while.
> >>>
> >>> Question, anyone successful in deploying RTPProxy to a dockerized
> environment? Preferably to a Kubernetes managed environment.
> >>>
> >>> Please Help Team :)
> >>>
> >>> Kind Regards,
> >>> Terrance
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
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Re: [OpenSIPS-Users] RTPProxy Docker Image

[Saint Michael]

Opensips+ RTPProxy only works fine with plain LXC containers,
privileged, which basically have access to all the resources of the
box.
That is the model I use with great success.

On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff  wrote:
>
> Hello Terrance,
> I wouldn't really recommend this. RTPProxy is going to use a lot of ports in 
> a very large range. That just doesn't work great in docker, but even worse in 
> K8S.
>
> I personally would put the RTPProxy outside of K8S. While you might be able 
> to get it to work, you are likely going against some basic design concepts in 
> containerization. I feel like the tech should propel the solution and not be 
> a hindrance to it. In this case, I'm not sure that K8S is buying you anything 
> of value, but instead creating architectural challenges.
>
> I'd love to hear feedback or experiences from others. There's always 
> something to learn :)
> -Brett
>
> On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor  wrote:
>>
>> Was it something I said?
>>
>> Terrance
>>
>> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor  wrote:
>>>
>>> Hello Everyone,
>>>
>>> Wow! Blast from the past... I am a long time member of this list, been a 
>>> while.
>>>
>>> Question, anyone successful in deploying RTPProxy to a dockerized 
>>> environment? Preferably to a Kubernetes managed environment.
>>>
>>> Please Help Team :)
>>>
>>> Kind Regards,
>>> Terrance
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
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Re: [OpenSIPS-Users] RTPProxy Docker Image

[Brett Nemeroff]

Hello Terrance,
I wouldn't really recommend this. RTPProxy is going to use a lot of ports
in a very large range. That just doesn't work great in docker, but even
worse in K8S.

I personally would put the RTPProxy outside of K8S. While you might be able
to get it to work, you are likely going against some basic design concepts
in containerization. I feel like the tech should propel the solution and
not be a hindrance to it. In this case, I'm not sure that K8S is buying you
anything of value, but instead creating architectural challenges.

I'd love to hear feedback or experiences from others. There's always
something to learn :)
-Brett

On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor  wrote:

> Was it something I said?
>
> Terrance
>
> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor 
> wrote:
>
>> Hello Everyone,
>>
>> Wow! Blast from the past... I am a long time member of this list, been a
>> while.
>>
>> Question, anyone successful in deploying RTPProxy to a dockerized
>> environment? Preferably to a Kubernetes managed environment.
>>
>> Please Help Team :)
>>
>> Kind Regards,
>> Terrance
>>
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Re: [OpenSIPS-Users] RTPProxy Docker Image

[Terrance Devor]

Was it something I said?

Terrance

On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor  wrote:

> Hello Everyone,
>
> Wow! Blast from the past... I am a long time member of this list, been a
> while.
>
> Question, anyone successful in deploying RTPProxy to a dockerized
> environment? Preferably to a Kubernetes managed environment.
>
> Please Help Team :)
>
> Kind Regards,
> Terrance
>
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[OpenSIPS-Users] RTPProxy Docker Image

[Terrance Devor]

Hello Everyone,

Wow! Blast from the past... I am a long time member of this list, been a
while.

Question, anyone successful in deploying RTPProxy to a dockerized
environment? Preferably to a Kubernetes managed environment.

Please Help Team :)

Kind Regards,
Terrance
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Re: [OpenSIPS-Users] Rtpproxy with mhomed option

[Mario San Vicente]

Hello John,

I have missed your answer as it was on the Spam folder.   I got it working
with rtpproxy_engage(""), but now I can see that the trick was passing the
parameter with double quotes.


Thank you for your valuable help!
Mario

On Mon, Feb 22, 2021 at 7:03 AM John Quick 
wrote:

> Hi Mario,
>
> I think you are talking about bridged mode for rtpproxy.
> If so, you will not need to start the rtpproxy daemon with -A, because that
> is where you have just one interface and you want to masquerade it as a
> different address. For example if your server is behind NAT, then -l is the
> actual host interface address and -A is the external IP address on the WAN
> port of the NAT router.
>
> For bridging mode, you have to specify two addresses with the -l parameter.
> This is what I have in my notes:
> -l is the listen address or pair of addresses when used in bridging mode
> (LAN then WAN)
> e.g.-l 123.45.67.89 or-l 192.168.4.102/123.45.67.89
>
> Also, have you tried putting the parameters inside double quotes when
> calling rtpproxy_answer()?
> This was a requirement in v2.4.x
> I believe they tried to remove the requirement for quotes around parameters
> in v3, but maybe they missed it in a few places.
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
>

-- 
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Re: [OpenSIPS-Users] Rtpproxy with mhomed option

[John Quick]

Hi Mario,

I think you are talking about bridged mode for rtpproxy.
If so, you will not need to start the rtpproxy daemon with -A, because that
is where you have just one interface and you want to masquerade it as a
different address. For example if your server is behind NAT, then -l is the
actual host interface address and -A is the external IP address on the WAN
port of the NAT router.

For bridging mode, you have to specify two addresses with the -l parameter.
This is what I have in my notes:
-l is the listen address or pair of addresses when used in bridging mode
(LAN then WAN)
e.g.-l 123.45.67.89 or-l 192.168.4.102/123.45.67.89

Also, have you tried putting the parameters inside double quotes when
calling rtpproxy_answer()?
This was a requirement in v2.4.x
I believe they tried to remove the requirement for quotes around parameters
in v3, but maybe they missed it in a few places.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk



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[OpenSIPS-Users] Rtpproxy with mhomed option

[Mario San Vicente]

Hi Everyone,

I have been testing opensips with rtpproxy and it works really well on a
single interface server and its correct nating. But now that i want to run
a configuration with a wan and lan interface the ips at the SDP are not
correct.
I have seen in some blogs, that they use the flags (ie) to indicate if the
flow goes from internal to external,  but i am getting the following error:


opensips: CRITICAL:core:yyerror: parse error in
/usr/local/etc/opensips/opensips.cfg:520:22-23: bad arguments for command


some config related:


version: opensips 3.1.0
mhomed=1
socket=udp:x.x.x.x:5060   # external
socket=udp:y.y.y.y:5060  # internal

loadmodule "rtpproxy.so"
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:2123")
modparam("rtpproxy", "rtpproxy_autobridge", 1)


route[ rtpproxy _offer] {
rtpproxy_offer(ei);

}

route[rtpproxy_answer] {
rtpproxy_answer(ei);

}

NOTE: if a remove the flags (ei) , the dialog works, but with wrong media ip


For running rtpproxy

rtpproxy -l y.y.y.y -A x.x.x.x -m 1 -M 2 -d DBUG:LOG_LOCAL0 -s udp:
127.0.0.1:2123 -F


Thank you in advance
-- 
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Re: [OpenSIPS-Users] RTPproxy for OpenSIPS 3.1 on Debian 10.7 (How To)

[Rick McGill - ₪]

Dear Eugene,

Thank you very much for the links and pointing me in the right direction.
None of the online installation guides I had found for OpenSIPs even mentioned 
anything about RTPproxy.
The only reason I know about it was because of the ERROR's I found in the 
OpenSIPS log and Syslog.

Anyway I'm pointed in the right direction now by your email and will attempt 
this RTPproxy install today.


Rick McGill – CEO
r...@netrovoip.com | r...@netropolitanworks.com 
Thailand: +66-2105-4262  x1001  |   USA: +1-737-237-2030   |Mobile: 
+66-85557-3000
Support:: +66-97047-2000  |  SKYPE & LINE ID:  NetroVOIP  | 
supp...@netrovoip.com
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Provider
--
   

-Original Message-
From: Eugene Christensen  
Sent: Thursday, February 11, 2021 11:21 PM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] RTPproxy for OpenSIPS 3.1 on Debian 10.7 (How To)

Hi Rick.

I'm also new to OpenSIPS so take my suggestion with a grain of salt.

I used the information found here to install rtpproxy on my Debian system.
https://computingforgeeks.com/how-to-install-rtpproxy-from-source-on-ubuntu-linux/

Once installed, I have been running it manually for the time being instead of 
creating it as a service.  At some point, if I stick with RTPProxy, I'm sure 
I'll go the route of using the service.

Don't forget to configure the opensips.cfg to have the listen port setup for 
rtpproxy to match the port that you set rtpproxy to use.

Here are a couple of other links that I looked at in case you want to review 
some alternative options.
https://www.rtpproxy.org/doc/master/user_manual.html#MAKESRC

https://subscription.packtpub.com/book/networking_and_servers/9781849510745/9/ch09lvl1sec90/rtp-proxy-installation-and-configuration

Good luck.

Eugene Christensen.

CONFIDENTIALITY NOTICE. This e-mail transmission, and any documents, files or 
previous e-mail messages attached to it, may contain confidential and 
proprietary information. If you are not the intended recipient, or a person 
responsible for delivering it to the intended recipient, you are hereby 
notified that any disclosure, copying, distribution or use of any of the 
information contained in or attached to this message is STRICTLY PROHIBITED. If 
you have received this transmission in error, please immediately notify me by 
reply e-mail at echristen...@sorenson.com or by telephone at +1 (801) 287-9419, 
and destroy the original transmission and its attachments without reading them 
or saving them to disk.

-Original Message-
From: Users  On Behalf Of Rick McGill - ? 
Sent: Thursday, February 11, 2021 8:56 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] RTPproxy for OpenSIPS 3.1 on Debian 10.7 (How To)

[EXTERNAL] 

Dear OpenSIPS Community,

Yeah I'm a newby to OpenSIPs app specifically and I just realized that I need a 
separate install for RTPproxy for my OpenSIPS.

My question is there a repo for that or do I need to download and install it 
manually?
Also I cannot find and good material for how to install so if anyone knows of a 
site for that it would be helpful.

I'm have a fresh install of OpenSIPs 3.1 on Debian 10.7   and planning to
install the RTPproxy on the same machine.   



Rick McGill – CEO
r...@netrovoip.com | r...@netropolitanworks.com 
Thailand: +66-2105-4262  x1001  |   USA: +1-737-237-2030   |Mobile:
+66-85557-3000
Support:: +66-97047-2000  |  SKYPE & LINE ID:  NetroVOIP  | 
supp...@netrovoip.com
  ₪  www.NetroVOIP.com  Telecommunications / Video Consulting & Solutions 
Provider

--   

-Original Message-
From: Rick McGill - ₪ 
Sent: Thursday, February 11, 2021 2:27 PM
To: 'users@lists.opensips.org' 
Subject: OpenSIPs 3.1 New install ERROR in Logs rtpproxy

Dear OpenSIPs Community,

I have a fresh install of OpenSIPs 3.1 on Debian 10.7 I have done the basic 
install but it is yet to be configured with any gateways and only has a few 
test users setup on it.
I'm new to OpenSIPs.

The ERROR issue I'm seeing in the log files seems related to rtpproxy.
I’m not familiar yet with OpenSIPs so if someone could at least point in the 
right direction to start to solve this error it might save me a lot of time 
Googling.

-
Feb 11 13:32:03 sip /usr/sbin/opensips[1704]: WARNING:rtpproxy:rtpp_test:
support for RTP proxy  has been disabled temporarily Feb 
11 13

Re: [OpenSIPS-Users] RTPproxy for OpenSIPS 3.1 on Debian 10.7 (How To)

[Eugene Christensen]

Hi Rick.

I'm also new to OpenSIPS so take my suggestion with a grain of salt.

I used the information found here to install rtpproxy on my Debian system.
https://computingforgeeks.com/how-to-install-rtpproxy-from-source-on-ubuntu-linux/

Once installed, I have been running it manually for the time being instead of 
creating it as a service.  At some point, if I stick with RTPProxy, I'm sure 
I'll go the route of using the service.

Don't forget to configure the opensips.cfg to have the listen port setup for 
rtpproxy to match the port that you set rtpproxy to use.

Here are a couple of other links that I looked at in case you want to review 
some alternative options.
https://www.rtpproxy.org/doc/master/user_manual.html#MAKESRC

https://subscription.packtpub.com/book/networking_and_servers/9781849510745/9/ch09lvl1sec90/rtp-proxy-installation-and-configuration

Good luck.

Eugene Christensen.

CONFIDENTIALITY NOTICE. This e-mail transmission, and any documents, files or 
previous e-mail messages attached to it, may contain confidential and 
proprietary information. If you are not the intended recipient, or a person 
responsible for delivering it to the intended recipient, you are hereby 
notified that any disclosure, copying, distribution or use of any of the 
information contained in or attached to this message is STRICTLY PROHIBITED. If 
you have received this transmission in error, please immediately notify me by 
reply e-mail at echristen...@sorenson.com or by telephone at +1 (801) 287-9419, 
and destroy the original transmission and its attachments without reading them 
or saving them to disk.

-Original Message-
From: Users  On Behalf Of Rick McGill - ? 
Sent: Thursday, February 11, 2021 8:56 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] RTPproxy for OpenSIPS 3.1 on Debian 10.7 (How To)

[EXTERNAL] 

Dear OpenSIPS Community,

Yeah I'm a newby to OpenSIPs app specifically and I just realized that I need a 
separate install for RTPproxy for my OpenSIPS.

My question is there a repo for that or do I need to download and install it 
manually?
Also I cannot find and good material for how to install so if anyone knows of a 
site for that it would be helpful.

I'm have a fresh install of OpenSIPs 3.1 on Debian 10.7   and planning to
install the RTPproxy on the same machine.   



Rick McGill – CEO
r...@netrovoip.com | r...@netropolitanworks.com 
Thailand: +66-2105-4262  x1001  |   USA: +1-737-237-2030   |Mobile:
+66-85557-3000
Support:: +66-97047-2000  |  SKYPE & LINE ID:  NetroVOIP  | 
supp...@netrovoip.com
  ₪  www.NetroVOIP.com  Telecommunications / Video Consulting & Solutions 
Provider

--   

-Original Message-
From: Rick McGill - ₪ 
Sent: Thursday, February 11, 2021 2:27 PM
To: 'users@lists.opensips.org' 
Subject: OpenSIPs 3.1 New install ERROR in Logs rtpproxy

Dear OpenSIPs Community,

I have a fresh install of OpenSIPs 3.1 on Debian 10.7 I have done the basic 
install but it is yet to be configured with any gateways and only has a few 
test users setup on it.
I'm new to OpenSIPs.

The ERROR issue I'm seeing in the log files seems related to rtpproxy.
I’m not familiar yet with OpenSIPs so if someone could at least point in the 
right direction to start to solve this error it might save me a lot of time 
Googling.

-
Feb 11 13:32:03 sip /usr/sbin/opensips[1704]: WARNING:rtpproxy:rtpp_test:
support for RTP proxy  has been disabled temporarily Feb 
11 13:32:03 sip /usr/sbin/opensips[1703]:
ERROR:rtpproxy:send_rtpp_command: can't send (#2 iovec buffers) command to a 
RTP proxy (111:Connection refused) Feb 11 13:32:03 sip
/usr/sbin/opensips[1703]: ERROR:rtpproxy:send_rtpp_command: proxy 
 does not respond, disable it Feb 11 13:32:03 sip
/usr/sbin/opensips[1703]: WARNING:rtpproxy:rtpp_test: can't get version of the 
RTP proxy
-




Rick McGill – CEO
r...@netrovoip.com | r...@netropolitanworks.com 
Thailand: +66-2105-4262  x1001  |   USA: +1-737-237-2030   |Mobile:
+66-85557-3000
Support:: +66-97047-2000  |  SKYPE & LINE ID:  NetroVOIP  | 
supp...@netrovoip.com
  ₪  www.NetroVOIP.com  Telecommunications / Video Consulting & Solutions 
Provider

--   




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U

[OpenSIPS-Users] RTPproxy for OpenSIPS 3.1 on Debian 10.7 (How To)

[Rick McGill - ₪]

Dear OpenSIPS Community,

Yeah I'm a newby to OpenSIPs app specifically and I just realized that I
need a separate install for RTPproxy for my OpenSIPS.

My question is there a repo for that or do I need to download and install it
manually?
Also I cannot find and good material for how to install so if anyone knows
of a site for that it would be helpful.

I'm have a fresh install of OpenSIPs 3.1 on Debian 10.7   and planning to
install the RTPproxy on the same machine.   



Rick McGill – CEO
r...@netrovoip.com | r...@netropolitanworks.com 
Thailand: +66-2105-4262  x1001  |   USA: +1-737-237-2030   |Mobile:
+66-85557-3000
Support:: +66-97047-2000  |  SKYPE & LINE ID:  NetroVOIP  |
supp...@netrovoip.com
  ₪  www.NetroVOIP.com  Telecommunications / Video Consulting & Solutions
Provider

--   

-Original Message-
From: Rick McGill - ₪  
Sent: Thursday, February 11, 2021 2:27 PM
To: 'users@lists.opensips.org' 
Subject: OpenSIPs 3.1 New install ERROR in Logs rtpproxy

Dear OpenSIPs Community,

I have a fresh install of OpenSIPs 3.1 on Debian 10.7 I have done the basic
install but it is yet to be configured with any gateways and only has a few
test users setup on it.
I'm new to OpenSIPs.

The ERROR issue I'm seeing in the log files seems related to rtpproxy.
I’m not familiar yet with OpenSIPs so if someone could at least point in the
right direction to start to solve this error it might save me a lot of time
Googling.

-
Feb 11 13:32:03 sip /usr/sbin/opensips[1704]: WARNING:rtpproxy:rtpp_test:
support for RTP proxy  has been disabled temporarily
Feb 11 13:32:03 sip /usr/sbin/opensips[1703]:
ERROR:rtpproxy:send_rtpp_command: can't send (#2 iovec buffers) command to a
RTP proxy (111:Connection refused) Feb 11 13:32:03 sip
/usr/sbin/opensips[1703]: ERROR:rtpproxy:send_rtpp_command: proxy
 does not respond, disable it Feb 11 13:32:03 sip
/usr/sbin/opensips[1703]: WARNING:rtpproxy:rtpp_test: can't get version of
the RTP proxy
-




Rick McGill – CEO
r...@netrovoip.com | r...@netropolitanworks.com 
Thailand: +66-2105-4262  x1001  |   USA: +1-737-237-2030   |Mobile:
+66-85557-3000
Support:: +66-97047-2000  |  SKYPE & LINE ID:  NetroVOIP  |
supp...@netrovoip.com
  ₪  www.NetroVOIP.com  Telecommunications / Video Consulting & Solutions
Provider

--   




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Re: [OpenSIPS-Users] rtpproxy 'Memory State' 'status'

[solarmon]

Hi Tomi,

The "opensipsctl fifo rtpproxy_show" command does not give a 'status', but
a 'disabled' value - for example:

Set:: 0
node:: udp::7898 index=0 disabled=0 weight=1
recheck_ticks=0
node:: udp:  :7898 index=1 disabled=0 weight=1
recheck_ticks=0
node:: udp:  :7898 index=2 disabled=0 weight=1
recheck_ticks=0

Is this 'disabled' value meant to mean a config or running status?

I thought this 'disabled' value is meant to mean whether it has been
disabled in config - i.e. if you add "=0" to the URI address of the
configured rtpproxy node. This is not these same as whether the rtpproxy is
active/healthy - which is what "Memory State" in CP is meant to mean?

Thank you.

On Wed, 24 Jun 2020 at 17:15, Tomi Hakkarainen  wrote:

> Hi,
>
> I believe MU is just typo should be MI
>
> The ’status’ should also be retrieved differently not with the
>
> $ opensipsctl fifo rtpproxy_show
>
> command
>
> I think you can get the ’status’ directly from the database with SQL
> query.
>
> Tomi
>
>
> On 24. Jun 2020, at 16.33, solarmon  wrote:
>
> 
> Hi Bogdan-Andrei,
>
> There is only 'Memory State' in CP. There is no 'status' in CP.
>
> Sorry, what is 'MU'?
>
> I'm using opensips 2.4.x if that makes any difference.
>
> Thank you.
>
> On Wed, 24 Jun 2020 at 13:49, Bogdan-Andrei Iancu 
> wrote:
>
>> Hi,
>>
>> In CP, the `status` is the status from the SQL table and the `memory
>> status` is the status provided by the MU rtpproxy_show.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>>
>> On 6/24/20 12:16 PM, solarmon wrote:
>>
>> Hi,
>>
>> The command "opensipsctl fifo rtpproxy_show" does not return the 'status'
>> of the rtpproxy node.
>>
>> In OpenSIPS Control Panel, the RTPProxy table has a '*Memory State*'
>> column which seems to be the 'status' of that node.
>>
>> How can I get this 'Memory State' 'status' in command line form so that
>> it can be scripted? I want to be able to get a health status of the
>> rtpproxy from OpenSIPS point of view and graph it over a period of time.
>>
>> Thank you!
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> ___
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Re: [OpenSIPS-Users] rtpproxy 'Memory State' 'status'

[Tomi Hakkarainen]

Hi,

I believe MU is just typo should be MI

The ’status’ should also be retrieved differently not with the 
$ opensipsctl fifo rtpproxy_show
command

I think you can get the ’status’ directly from the database with SQL query. 

Tomi


On 24. Jun 2020, at 16.33, solarmon  wrote:


Hi Bogdan-Andrei,

There is only 'Memory State' in CP. There is no 'status' in CP.

Sorry, what is 'MU'?

I'm using opensips 2.4.x if that makes any difference.

Thank you.

> On Wed, 24 Jun 2020 at 13:49, Bogdan-Andrei Iancu  wrote:
> Hi,
> 
> In CP, the `status` is the status from the SQL table and the `memory status` 
> is the status provided by the MU rtpproxy_show.
> 
> Regards,
>  Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> 
>> On 6/24/20 12:16 PM, solarmon wrote:
>> Hi,
>> 
>> The command "opensipsctl fifo rtpproxy_show" does not return the 'status' of 
>> the rtpproxy node.
>> 
>> In OpenSIPS Control Panel, the RTPProxy table has a 'Memory State' column 
>> which seems to be the 'status' of that node.
>> 
>> How can I get this 'Memory State' 'status' in command line form so that it 
>> can be scripted? I want to be able to get a health status of the rtpproxy 
>> from OpenSIPS point of view and graph it over a period of time.
>> 
>> Thank you!
>> 
>> 
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
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Re: [OpenSIPS-Users] rtpproxy 'Memory State' 'status'

[solarmon]

Hi Bogdan-Andrei,

There is only 'Memory State' in CP. There is no 'status' in CP.

Sorry, what is 'MU'?

I'm using opensips 2.4.x if that makes any difference.

Thank you.

On Wed, 24 Jun 2020 at 13:49, Bogdan-Andrei Iancu 
wrote:

> Hi,
>
> In CP, the `status` is the status from the SQL table and the `memory
> status` is the status provided by the MU rtpproxy_show.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>
> On 6/24/20 12:16 PM, solarmon wrote:
>
> Hi,
>
> The command "opensipsctl fifo rtpproxy_show" does not return the 'status'
> of the rtpproxy node.
>
> In OpenSIPS Control Panel, the RTPProxy table has a '*Memory State*'
> column which seems to be the 'status' of that node.
>
> How can I get this 'Memory State' 'status' in command line form so that it
> can be scripted? I want to be able to get a health status of the rtpproxy
> from OpenSIPS point of view and graph it over a period of time.
>
> Thank you!
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
___
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Re: [OpenSIPS-Users] rtpproxy 'Memory State' 'status'

[Bogdan-Andrei Iancu]

Hi,

In CP, the `status` is the status from the SQL table and the `memory 
status` is the status provided by the MU rtpproxy_show.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com

On 6/24/20 12:16 PM, solarmon wrote:

Hi,

The command "opensipsctl fifo rtpproxy_show" does not return the 
'status' of the rtpproxy node.


In OpenSIPS Control Panel, the RTPProxy table has a '*Memory State*' 
column which seems to be the 'status' of that node.


How can I get this 'Memory State' 'status' in command line form so 
that it can be scripted? I want to be able to get a health status of 
the rtpproxy from OpenSIPS point of view and graph it over a period of 
time.


Thank you!

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[OpenSIPS-Users] rtpproxy 'Memory State' 'status'

[solarmon]

Hi,

The command "opensipsctl fifo rtpproxy_show" does not return the 'status'
of the rtpproxy node.

In OpenSIPS Control Panel, the RTPProxy table has a '*Memory State*' column
which seems to be the 'status' of that node.

How can I get this 'Memory State' 'status' in command line form so that it
can be scripted? I want to be able to get a health status of the rtpproxy
from OpenSIPS point of view and graph it over a period of time.

Thank you!
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Re: [OpenSIPS-Users] [RTPproxy] catch_dtmf module has landed in rtpproxy/master

[Maxim Sobolev]

Thanks, Gohar! Very good questions:

1. Module relies on RFC2833 (DTMF Events). No in-band decoding is
implemented at the moment.

2. Module is observer-only for now. It doesn't try to block or in any way
alter the RTP stream being forwarded. Hovewer, that might be something to
consider as a next step, in fact it might be almost trivial to replace the
actual digits with some junk.

Our goal with this initial release was to provide something that could be
immediately useful and unblock OpenSIPS work in this area. Big part of the
effort went to refine and build up interfaces on rtpproxy side to make
future development easier for this functionality as well as allow us and
others implementing various packet-processing options (i.e. bridge_srtp
certainly high on priority list). Also, the module infrastructure is
decoupled from the core, so it should allow using all kinds of crazy
external code libraries (DSP, voice recognition etc), without compromising
our core integrity when the functionality is not used.

-Max

On Thu., May 14, 2020, 7:25 p.m. Gohar Ahmed,  wrote:

> Hi,
> That's big news indeed. Quick question though, is it capable of capturing
> inband DTMF and possibly drop them ?
>
> Looking forward for next episode of SIP chronicles.
>
> Best Regards,
> Gohar Ahmed
>
> On Thu., May 14, 2020, 9:10 p.m. Maxim Sobolev, 
> wrote:
>
>> Hi Razvan & OpenSIPS-users,
>>
>> This is just a quick heads up about catch_dtmf functionality being
>> available in the rtpproxy/master effective immediately. It is needed for
>> the DTMF call control feature in OpenSIPS 3.1. The only difference between
>> it and rtpp_2_1_dtmf code is that in order to enable the feature in master
>> one needs to load catch_dtmf module either by using "--dso
>> /some/where/rtpp_catch_dtmf.so" or by providing configuration file
>> with catch_dtmf section in modules:
>>
>> modules {
>> [...]
>> catch_dtmf {
>> load = /some/where/rtpp_catch_dtmf.so
>> }
>> }
>>
>> The module code has been developed in collab with Razvan + sponsored by
>> the OpenSIPS Solutions and comes with the test case providing 95%
>> coverage*. Any feedback is highly appeciated, as usually, happy DTMF'ing!
>>
>> -Max
>> *)
>> https://coveralls.io/builds/30800066/source?filename=modules/catch_dtmf/rtpp_catch_dtmf.c
>>
>> --
>> You received this message because you are subscribed to the Google Groups
>> "rtpproxy" group.
>> To unsubscribe from this group and stop receiving emails from it, send an
>> email to rtpproxy+unsubscr...@googlegroups.com.
>> To view this discussion on the web visit
>> https://groups.google.com/d/msgid/rtpproxy/CAH7qZfuucLr2TYrYFT6X%2B9WL%3D7xay7pJsExUxAhp5gb4UL5%2Bsg%40mail.gmail.com
>> 
>> .
>>
>
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Re: [OpenSIPS-Users] rtpproxy module not supporting valid payload types

[Maxim Sobolev]

Yes, for sure. As long as the transport is UDP based, the RTPProxy would
just work. The change should be trivial, you can get it fixed locally, test
and then open a pull request against opensips repo.

-Max


On Thu., Apr. 2, 2020, 11:43 a.m. Robert Dyck,  wrote:

> Regarding opensips-3.0
>
> Use case - webrtc client behind NAT
>
>
>
> The rtpproxy module emitted the error message "can't extract media port
> from the message" ( by the way, very misleading ). In reality
> extract_mediainfo fails because it could not find a supported payload type
> in the media description. The payload type in question is
> "UDP/TLS/RTP/SAVPF".
>
>
>
> RFC 5764 section 8 introduces four more RTP types.
>
> DCCP/TLS/RTP/SAVP and SAVPF
>
> UDP/TLS/RTP/SAVP and SAVPF
>
>
>
> Should rtpproxy.c be extended to support these additional RTP types?
>
>
>
> Thank you, Rob
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[OpenSIPS-Users] rtpproxy module not supporting valid payload types

[Robert Dyck]

Regarding opensips-3.0
Use case - webrtc client behind NAT

The rtpproxy module emitted the error message "can't extract media port from 
the message" ( by the way, very misleading ). In reality extract_mediainfo 
fails 
because it could not find a supported payload type in the media description. 
The 
payload type in question is "UDP/TLS/RTP/SAVPF".

RFC 5764 section 8 introduces four more RTP types.
DCCP/TLS/RTP/SAVP and SAVPF
UDP/TLS/RTP/SAVP and SAVPF

Should rtpproxy.c be extended to support these additional RTP types?

Thank you, Rob
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Re: [OpenSIPS-Users] rtpproxy status health checking

[Callum Guy]

Hi Solarmon,

I can't comment on the exact behaviour internally regarding the ticks value
however I thought I could share it as I see it from a user perspective.

The relevant settings I use are as follows:

modparam("rtpproxy", "rtpproxy_disable_tout", 60)
modparam("rtpproxy", "rtpproxy_timeout", "0.2")
modparam("rtpproxy", "rtpproxy_retr", 4)

My understanding is that it causes the following behaviour:

   1. OpenSIPs selects a RTPProxy instance from the pool and sends it the
   request
   2. RTPProxy instance does not respond within 0.2s
   3. This is repeated up to 4 times (*I am not sure if the repeats happen
   for the same dialogs or for other dialogs, I would assume that OpenSIPs
   will just try another instance immediately and increment the failure count*
   )
   4. If 4 successive requests for a given instance time out the node is
   flagged as disabled for 60 seconds and the other instances are used
   5. After 60 seconds the unresponsive instance is re-added to the pool

So regarding health checks, I do not believe this is a separate "heartbeat
system" instead it is a matter of tracking communication failures and using
this to disable unresponsive instances as per your configuration.

I hope that helps,

Callum


On Tue, 20 Aug 2019 at 09:55, solarmon  wrote:

> Hi,
>
> Just an update/correction. I notice that when I make a call through
> opensips, the recheck_ticks value does reduce slightly to 928855, but it
> seems to stay at that for subsequent calls. In Control Panel. The memory
> state does turn to Green.
>
> # opensipsctl fifo rtpproxy_show
> Set:: 0
> node:: udp: index=0 disabled=0 weight=1
> recheck_ticks=928855
> node:: udp: index=1 disabled=0 weight=1 recheck_ticks=0
> node:: udp: index=2 disabled=0 weight=1 recheck_ticks=0
>
> Please can you explain what recheck_ticks is and does and how it relates
> to any timeouts and health checks.
>
> On Tue, 20 Aug 2019 at 09:16, solarmon  wrote:
>
>> Hi Razvan,
>>
>> I'm using opensips 2.4.6 (x86_64/linux) so I don't think opensips-cli is
>> available?
>>
>> I'm using opensipsctl to show the rtpproxy status.
>>
>> This is the output of the command after I have turned off rtpproxy with
>> index 0:
>>
>> # opensipsctl fifo rtpproxy_show
>> Set:: 0
>> node:: udp: index=0 disabled=0 weight=1 recheck_ticks=0
>> node:: udp:index=1 disabled=0 weight=1
>> recheck_ticks=0
>> node:: udp:index=2 disabled=0 weight=1
>> recheck_ticks=0
>>
>> OpenSIPS Control Panel shows the same - the status does not change.
>>
>> When it does change, based on my quick testing, is:
>>
>> 1. On Reload
>> 2. When the is a call setup goes through it.
>>
>> I'm expecting the health/status to be checked on a regular basis, so as
>> to provide for early detection of failure.
>>
>> After I perform a reload (of rtpproxy config):
>>
>> # opensipsctl fifo rtpproxy_show
>> Set:: 0
>> node:: udp:index=0 disabled=1 weight=1
>> recheck_ticks=926858
>> node:: udp:index=1 disabled=0 weight=1
>> recheck_ticks=0
>> node:: udp:1index=2 disabled=0 weight=1
>> recheck_ticks=0
>>
>> The status of rtpproxy node with index 0 shows a value (926858) for
>> recheck_ticks. However this value never changes - it always shows 926858.
>>
>> On Mon, 19 Aug 2019 at 15:25, Răzvan Crainea  wrote:
>>
>>> Hi, Solarmon!
>>>
>>> The parameter you should use is exactly the one you are using,
>>> rtpproxy_disable_tout[1]. That parameter says that after OpenSIPS
>>> detects the node as being down, it re-tries to send them requests after
>>> 20 seconds (according to your configuration).
>>> Are you checking the rtpproxy status using the `opensips-cli mi`? Does
>>> the disable timeout change? If not, what's the output of the status
>>> command?
>>>
>>> [1]
>>>
>>> https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#param_rtpproxy_disable_tout
>>>
>>> Best regards,
>>> Răzvan
>>>
>>> On 8/13/19 4:14 PM, solarmon wrote:
>>> > Hi,
>>> >
>>> > Can somebody clarify when the rtpproxy status and health checks are
>>> done
>>> > and what configuration is required.
>>> >
>>> > I am finding that the status/health of an rtprpoxy node is only
>>> > done/checked during opensips startup or rtpproxy module config reload.
>>> > If the rtprpoxy node goes down or comes back up, the status indicated
>>> by
>>> > opensips for that rtpproxy does not change until another restart or
>>> > reload is done.
>>> >
>>> > My rtpproxy module config is as below:
>>> >
>>> > loadmodule "rtpproxy.so"
>>> > modparam("rtpproxy", "db_url",
>>> > "mysql://@127.0.0.1:3306/opensips
>>> > ")
>>> > modparam("rtpproxy", "rtpproxy_disable_tout", 20)
>>> >
>>> > Thank you in advance for any help provided.
>>> >
>>> > ___
>>> > Users mailing list
>>> > Users@lists.opensips.org
>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> >
>>>
>>> --
>>> Răzvan Crainea
>>> OpenSIPS C

Re: [OpenSIPS-Users] rtpproxy status health checking

[solarmon]

Hi,

Just an update/correction. I notice that when I make a call through
opensips, the recheck_ticks value does reduce slightly to 928855, but it
seems to stay at that for subsequent calls. In Control Panel. The memory
state does turn to Green.

# opensipsctl fifo rtpproxy_show
Set:: 0
node:: udp: index=0 disabled=0 weight=1
recheck_ticks=928855
node:: udp: index=1 disabled=0 weight=1 recheck_ticks=0
node:: udp: index=2 disabled=0 weight=1 recheck_ticks=0

Please can you explain what recheck_ticks is and does and how it relates to
any timeouts and health checks.

On Tue, 20 Aug 2019 at 09:16, solarmon  wrote:

> Hi Razvan,
>
> I'm using opensips 2.4.6 (x86_64/linux) so I don't think opensips-cli is
> available?
>
> I'm using opensipsctl to show the rtpproxy status.
>
> This is the output of the command after I have turned off rtpproxy with
> index 0:
>
> # opensipsctl fifo rtpproxy_show
> Set:: 0
> node:: udp: index=0 disabled=0 weight=1 recheck_ticks=0
> node:: udp:index=1 disabled=0 weight=1 recheck_ticks=0
> node:: udp:index=2 disabled=0 weight=1 recheck_ticks=0
>
> OpenSIPS Control Panel shows the same - the status does not change.
>
> When it does change, based on my quick testing, is:
>
> 1. On Reload
> 2. When the is a call setup goes through it.
>
> I'm expecting the health/status to be checked on a regular basis, so as to
> provide for early detection of failure.
>
> After I perform a reload (of rtpproxy config):
>
> # opensipsctl fifo rtpproxy_show
> Set:: 0
> node:: udp:index=0 disabled=1 weight=1
> recheck_ticks=926858
> node:: udp:index=1 disabled=0 weight=1 recheck_ticks=0
> node:: udp:1index=2 disabled=0 weight=1
> recheck_ticks=0
>
> The status of rtpproxy node with index 0 shows a value (926858) for
> recheck_ticks. However this value never changes - it always shows 926858.
>
> On Mon, 19 Aug 2019 at 15:25, Răzvan Crainea  wrote:
>
>> Hi, Solarmon!
>>
>> The parameter you should use is exactly the one you are using,
>> rtpproxy_disable_tout[1]. That parameter says that after OpenSIPS
>> detects the node as being down, it re-tries to send them requests after
>> 20 seconds (according to your configuration).
>> Are you checking the rtpproxy status using the `opensips-cli mi`? Does
>> the disable timeout change? If not, what's the output of the status
>> command?
>>
>> [1]
>>
>> https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#param_rtpproxy_disable_tout
>>
>> Best regards,
>> Răzvan
>>
>> On 8/13/19 4:14 PM, solarmon wrote:
>> > Hi,
>> >
>> > Can somebody clarify when the rtpproxy status and health checks are
>> done
>> > and what configuration is required.
>> >
>> > I am finding that the status/health of an rtprpoxy node is only
>> > done/checked during opensips startup or rtpproxy module config reload.
>> > If the rtprpoxy node goes down or comes back up, the status indicated
>> by
>> > opensips for that rtpproxy does not change until another restart or
>> > reload is done.
>> >
>> > My rtpproxy module config is as below:
>> >
>> > loadmodule "rtpproxy.so"
>> > modparam("rtpproxy", "db_url",
>> > "mysql://@127.0.0.1:3306/opensips
>> > ")
>> > modparam("rtpproxy", "rtpproxy_disable_tout", 20)
>> >
>> > Thank you in advance for any help provided.
>> >
>> > ___
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>>
>> --
>> Răzvan Crainea
>> OpenSIPS Core Developer
>>http://www.opensips-solutions.com
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
___
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Re: [OpenSIPS-Users] rtpproxy status health checking

[solarmon]

Hi Razvan,

I'm using opensips 2.4.6 (x86_64/linux) so I don't think opensips-cli is
available?

I'm using opensipsctl to show the rtpproxy status.

This is the output of the command after I have turned off rtpproxy with
index 0:

# opensipsctl fifo rtpproxy_show
Set:: 0
node:: udp: index=0 disabled=0 weight=1 recheck_ticks=0
node:: udp:index=1 disabled=0 weight=1 recheck_ticks=0
node:: udp:index=2 disabled=0 weight=1 recheck_ticks=0

OpenSIPS Control Panel shows the same - the status does not change.

When it does change, based on my quick testing, is:

1. On Reload
2. When the is a call setup goes through it.

I'm expecting the health/status to be checked on a regular basis, so as to
provide for early detection of failure.

After I perform a reload (of rtpproxy config):

# opensipsctl fifo rtpproxy_show
Set:: 0
node:: udp:index=0 disabled=1 weight=1
recheck_ticks=926858
node:: udp:index=1 disabled=0 weight=1 recheck_ticks=0
node:: udp:1index=2 disabled=0 weight=1 recheck_ticks=0

The status of rtpproxy node with index 0 shows a value (926858) for
recheck_ticks. However this value never changes - it always shows 926858.

On Mon, 19 Aug 2019 at 15:25, Răzvan Crainea  wrote:

> Hi, Solarmon!
>
> The parameter you should use is exactly the one you are using,
> rtpproxy_disable_tout[1]. That parameter says that after OpenSIPS
> detects the node as being down, it re-tries to send them requests after
> 20 seconds (according to your configuration).
> Are you checking the rtpproxy status using the `opensips-cli mi`? Does
> the disable timeout change? If not, what's the output of the status
> command?
>
> [1]
>
> https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#param_rtpproxy_disable_tout
>
> Best regards,
> Răzvan
>
> On 8/13/19 4:14 PM, solarmon wrote:
> > Hi,
> >
> > Can somebody clarify when the rtpproxy status and health checks are done
> > and what configuration is required.
> >
> > I am finding that the status/health of an rtprpoxy node is only
> > done/checked during opensips startup or rtpproxy module config reload.
> > If the rtprpoxy node goes down or comes back up, the status indicated by
> > opensips for that rtpproxy does not change until another restart or
> > reload is done.
> >
> > My rtpproxy module config is as below:
> >
> > loadmodule "rtpproxy.so"
> > modparam("rtpproxy", "db_url",
> > "mysql://@127.0.0.1:3306/opensips
> > ")
> > modparam("rtpproxy", "rtpproxy_disable_tout", 20)
> >
> > Thank you in advance for any help provided.
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>http://www.opensips-solutions.com
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] rtpproxy status health checking

[Răzvan Crainea]

Hi, Solarmon!

The parameter you should use is exactly the one you are using, 
rtpproxy_disable_tout[1]. That parameter says that after OpenSIPS 
detects the node as being down, it re-tries to send them requests after 
20 seconds (according to your configuration).
Are you checking the rtpproxy status using the `opensips-cli mi`? Does 
the disable timeout change? If not, what's the output of the status command?


[1] 
https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#param_rtpproxy_disable_tout


Best regards,
Răzvan

On 8/13/19 4:14 PM, solarmon wrote:

Hi,

Can somebody clarify when the rtpproxy status and health checks are done 
and what configuration is required.


I am finding that the status/health of an rtprpoxy node is only 
done/checked during opensips startup or rtpproxy module config reload. 
If the rtprpoxy node goes down or comes back up, the status indicated by 
opensips for that rtpproxy does not change until another restart or 
reload is done.


My rtpproxy module config is as below:

loadmodule "rtpproxy.so"
modparam("rtpproxy", "db_url", 
"mysql://@127.0.0.1:3306/opensips 
")

modparam("rtpproxy", "rtpproxy_disable_tout", 20)

Thank you in advance for any help provided.

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--
Răzvan Crainea
OpenSIPS Core Developer
  http://www.opensips-solutions.com

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[OpenSIPS-Users] rtpproxy status health checking

[solarmon]

Hi,

Can somebody clarify when the rtpproxy status and health checks are done
and what configuration is required.

I am finding that the status/health of an rtprpoxy node is only
done/checked during opensips startup or rtpproxy module config reload. If
the rtprpoxy node goes down or comes back up, the status indicated by
opensips for that rtpproxy does not change until another restart or reload
is done.

My rtpproxy module config is as below:

loadmodule "rtpproxy.so"
modparam("rtpproxy", "db_url", "mysql://@
127.0.0.1:3306/opensips")
modparam("rtpproxy", "rtpproxy_disable_tout", 20)

Thank you in advance for any help provided.
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Re: [OpenSIPS-Users] RTPProxy No Audio on Outbound Calls

[John Quick]

Mark,

You can detect if the INVITE came from your Asterisk by testing the $si
pseudo-variable.
That will allow you to identify the direction of the call. I usually set a
flag for this purpose. For example:
If ($si == "my.ast.er.isk")
setflag(DIR_OUT);

At the point where you engage the rtpproxy, you will then be able to reverse
the internal/external parameters for the function call depending on the
direction of the call

If (isflagset(DIR_OUT)) {
rtpproxy_offer("corfei");
} else {
rtpproxy_offer("corfie");
}

The same flag should still be valid in the onreply handler where you can do
something similar. [Not sure if I have ie/ei the right way round in my
example].

That said, I'm not sure this topology is a good one to be using.
I would generally try to avoid having the media proxy behind NAT and also
using it in bridging mode - it makes life too complicated.

P.S. Looks like you sorted out the problems with the call to do_routing().

John Quick
Smartvox Limited



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[OpenSIPS-Users] RTPProxy No Audio on Outbound Calls

[Mark Farmer]

Hello everyone, all help gratefully received, I've been slogging away at
this for ages!

I have OpenSIPS 2.4.4 & RTPProxy behind 1:1 NAT's (different hosts).

RTPProxy runs so:
/usr/local/bin/rtpproxy -s unix:/var/run/rtpproxy/rtpproxy.sock -u rtpproxy
rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -s udp:10.96.16.58 7722 -l
10.96.0.58 10.98.0.58 -A ext.ip.addr.ess 10.98.0.58 -d DBUG LOG_LOCAL0 -m
1 -M 2

OpenSIPS is sitting between my provider & an Asterisk server which has
phones registered.

When I make calls 'Provider -> OpenSIPS/RTPProxy -> Asterisk -> Phone' all
is good, 2 way audio.
But when the call flows in the opposite direction, I get no audio since SDP
is the same as the 1st call.

How do I get it to reverse the rtpproxy_offer/answer flags?

These are the bits that handles it all:

route[RTPPROXY] {

if (is_method("BYE|CANCEL")) {
rtpproxy_unforce();
}

if (is_method("INVITE")) {
rtpproxy_offer("corwfie");
}
}

onreply_route[DROUTING] {

if (is_method("BYE|CANCEL")) {
sip_trace("tid","d");
rtpproxy_unforce();
}

if ($rs=~"(2[0-9][0-9])") {
rtpproxy_answer("corwfei");
}
}



-- 
Mark Farmer
farm...@gmail.com
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Re: [OpenSIPS-Users] RTPPROXY ENGAGEMENT

[Ovidiu Sas]

If you are experiencing double sdp re-wites it means that you are
engaging rtpproxy more then once.
Add some xlogs in your script where you engage rtpproxy and figure out
why it is engaged twice.

Regards,
Ovidiu Sas

On Fri, Jan 25, 2019 at 12:57 PM Mark Thomas  wrote:
>
> I have an issue and I don’t know how to resolve it. I’ve got a 486 route that 
> has to engage rtpproxy to bridge to another network. My problem is I have to 
> engage from UAC to UAS on the public side before it sends to voicemail. 
> Whenever I attempt to engage rtpproxy on the leg going to the voicemail 
> servers which are on load balancer it re-writes the sdp twice. I’ve battled 
> with the thing for a while now and could really use some assistance in 
> rectifying this issue.
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



-- 
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http://www.voipembedded.com

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[OpenSIPS-Users] RTPPROXY ENGAGEMENT

[Mark Thomas]

I have an issue and I don’t know how to resolve it. I’ve got a 486 route that 
has to engage rtpproxy to bridge to another network. My problem is I have to 
engage from UAC to UAS on the public side before it sends to voicemail. 
Whenever I attempt to engage rtpproxy on the leg going to the voicemail servers 
which are on load balancer it re-writes the sdp twice. I’ve battled with the 
thing for a while now and could really use some assistance in rectifying this 
issue. 

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[OpenSIPS-Users] RTPPROXY not starting on correct user account

[Steven Platt]

Greetings,

I am having trouble with RTPProxy bridging media in Opensips 2.3.5

Currently, it seems that RTPProxy is starting as user "root" and not user
"rtpproxy" as it should.
In etc/init.d/rtpproxy I have added the additional daemon options to load
the service as user "rtpproxy", but these seem to be ignored. Running
RTPProxy as root, gives me log errors and does not function.

DAEMON_OPTS="-l [server public ip] -s udp:127.0.0.1:7890 -u rtpproxy
rtpproxy -d DBUG:LOG_LOCAL0"

I can manually kill the existing rtpproxy process and start it again with
the switches to run in the correctly user. If I do this I get working media
on my LAN, but still nothing when connecting with a phone on cellular NAT -
even thought opensips sees RTP proxy sock available.

So I have 2 problems:
- RTPPROXY not starting as "rtpproxy" user
- RTPPROXY when working as normal, does not bridge media for cellular
device NAT.

# I later kill the PID for the RTPproxy instance running as root
root@opensips-server:~# ps aux | grep rtpproxy
rtpproxy  4458  0.0  0.0  93232  1596 ?Ssl  09:09   0:00
/usr/bin/rtpproxy -l [ server public ip] -s udp:127.0.0.1 7890 -u rtpproxy
rtpproxy -d DBUG LOG_LOCAL0
root  4784  0.0  0.0   4336  1624 ?Ss   09:12   0:00 /bin/sh
/etc/init.d/rtpproxy start
root  4787  0.0  0.0  19476  2196 ?S09:12   0:00
/usr/bin/rtpproxy -l [ server public ip] -s udp:127.0.0.1:7890 -u rtpproxy
rtpproxy -d DBUG LOG_LOCAL0
root  4831  0.0  0.0  12732  2136 pts/0S+   09:13   0:00 grep
rtpproxy

### RTP Proxy Config
PATH=/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin
NAME=rtpproxy
DESC="RTP relay"
DAEMON=/usr/bin/$NAME
USER=$NAME
GROUP=$USER
PIDFILE="/var/run/$NAME/$NAME.pid"
PIDFILE_DIR=`dirname $PIDFILE`
CONTROL_SOCK="unix:$PIDFILE_DIR/$NAME.sock"

test -x $DAEMON || exit 0
umask 002

. /lib/lsb/init-functions

# Include defaults if available
if [ -f /etc/default/$NAME ] ; then
. /etc/default/$NAME
fi

DAEMON_OPTS="-l [server public ip] -s udp:127.0.0.1:7890 -u rtpproxy
rtpproxy -d DBUG:LOG_LOCAL0"



 OpenSIPS Config
  NAT modules
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG")
modparam("nathelper", "sipping_from", "sip:pinger@127.0.0.1") #CUSTOMIZE ME
modparam("nathelper", "received_avp", "$avp(received_nh)")

loadmodule "rtpproxy.so"
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7890") # CUSTOMIZE ME


###Log from opensips look normal to me
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:handle_command:
received command "4443_3 VF 20081102"
Sep 27 14:25:42 opensips-server /usr/sbin/opensips[4452]:
INFO:rtpproxy:rtpp_test: rtp proxy  found, support for
it enabled
Sep 27 14:25:42 opensips-server /usr/sbin/opensips[4451]:
INFO:rtpproxy:rtpp_test: rtp proxy  found, support for
it enabled
Sep 27 14:25:42 opensips-server /usr/sbin/opensips[]:
INFO:rtpproxy:rtpp_test: rtp proxy  found, support for
it enabled
Sep 27 14:25:42 opensips-server /usr/sbin/opensips[4442]:
INFO:rtpproxy:rtpp_test: rtp proxy  found, support for
it enabled
Sep 27 14:25:42 opensips-server /usr/sbin/opensips[4461]:
INFO:rtpproxy:rtpp_test: rtp proxy  found, support for
it enabled
Sep 27 14:25:42 opensips-server /usr/sbin/opensips[4463]:
INFO:rtpproxy:rtpp_test: rtp proxy  found, support for
it enabled
Sep 27 14:25:42 opensips-server /usr/sbin/opensips[4464]:
INFO:rtpproxy:rtpp_test: rtp proxy  found, support for
it enabled
Sep 27 14:25:42 opensips-server /usr/sbin/opensips[4454]:
INFO:rtpproxy:rtpp_test: rtp proxy  found, support for
it enabled
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:doreply: sending reply
"4443_3 1#012"
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:handle_command:
received command "4433_3 VF 20081102"
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:doreply: sending reply
"4433_3 1#012"
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:handle_command:
received command "4434_1 VF 20050322"
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:doreply: sending reply
"4434_1 1#012"
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:handle_command:
received command "4437_1 VF 20050322"
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:doreply: sending reply
"4437_1 1#012"
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:handle_command:
received command "4442_0 V"
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:doreply: sending reply
"4442_0 20040107#012"
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:handle_command:
received command "4452_0 V"
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:doreply: sending reply
"4452_0 20040107#012"
Sep 27 14:25:42 opensips-server /usr/sbin/opensips[4437]:
WARNING:dialplan:dp_load_db: no data in the db
Sep 27 14:25:42 opensips-server rtpproxy[4424]: DBUG:handle_command:
received command "4451_0 V"
Sep 27 14:25:42 opensips-server opensips: I

Re: [OpenSIPS-Users] rtpproxy error messages

[John Quick]

Hello Razvan,

I have just seen your response to my question.
Thank you for explaining the rtpproxy error messages.
I will try the -L parameter as you suggest. I think this will be the answer.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk


> Hi, John!
>
> Those errors are reported by the RTPProxy server, and depending on the 
> error, they can be handled by opensips.
> In your case, the errors 71 and 72 are reported by RTPProxy when it can 
> not create UDP listeners for media. Usually these errors are triggered 
> when rtpproxy runs out of available sockets/file descriptors. You can 
> increase the number of sockets/file descriptors the RTPProxy server can 
> use using the -L parameter.
> Hope this helps.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Developer
> www.opensips-solutions.com


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Re: [OpenSIPS-Users] rtpproxy error messages

[Răzvan Crainea]

Hi, John!

Those errors are reported by the RTPProxy server, and depending on the 
error, they can be handled by opensips.
In your case, the errors 71 and 72 are reported by RTPProxy when it can 
not create UDP listeners for media. Usually these errors are triggered 
when rtpproxy runs out of available sockets/file descriptors. You can 
increase the number of sockets/file descriptors the RTPProxy server can 
use using the -L parameter.

Hope this helps.

Best regards,

Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com

On 01/16/2018 07:18 PM, John Quick wrote:

Hello,

Can anyone help me understand what might be causing the following error
messages to be reported by OpenSIPS?
2018-01-16 16:42:47  ERROR:rtpproxy:engage_rtp_proxy4_f: error forcing rtp
proxy
2018-01-16 16:42:48  ERROR:rtpproxy:force_rtp_proxy_body: unhandled rtpproxy
error: 71
2018-01-16 16:42:48  ERROR:rtpproxy:force_rtp_proxy_body: unhandled rtpproxy
error: 71
2018-01-16 16:42:49  ERROR:rtpproxy:force_rtp_proxy_body: unhandled rtpproxy
error: 72

Thanks.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk


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[OpenSIPS-Users] rtpproxy error messages

[John Quick]

Hello,

Can anyone help me understand what might be causing the following error
messages to be reported by OpenSIPS?
2018-01-16 16:42:47  ERROR:rtpproxy:engage_rtp_proxy4_f: error forcing rtp
proxy
2018-01-16 16:42:48  ERROR:rtpproxy:force_rtp_proxy_body: unhandled rtpproxy
error: 71
2018-01-16 16:42:48  ERROR:rtpproxy:force_rtp_proxy_body: unhandled rtpproxy
error: 71
2018-01-16 16:42:49  ERROR:rtpproxy:force_rtp_proxy_body: unhandled rtpproxy
error: 72

Thanks.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk


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Re: [OpenSIPS-Users] RTPProxy Timeout socket more than one SIP proxy

[Răzvan Crainea]

I don't think restarting RTPProxy is acceptable, because you will loose 
the ongoing RTP sessions.


Best regards,

Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com

On 01/15/2018 01:03 PM, Adrian Fretwell wrote:


Razvan,

Thankyou for clarifying this, I can work around it either with VIP or 
by restarting RTP Proxy with a different -n value.


Kind regards,

Adrian.

On 15/01/18 09:02, Răzvan Crainea wrote:

Hi, Adrian!

Unfortunately RTPProxy can send timeout notifications only to one 
timeout_socket - the one specified in the -n parameter. We did 
discuss with Maxim a while back the ability to support different 
timeout_sockets to different opensips instances, but that discussion 
didn't materialize yet.


However, for your setup, I guess you are already using a VIP(Virtual 
IP) that you share between the active and backup instance. An idea is 
to use the same VIP as the -n parameter - that way you will always 
send the timeout notification to the active OpenSIPS instance.


Let me know if that works for you.

Best regards,
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com
On 01/13/2018 03:26 PM, Adrian Fretwell wrote:

Hello & Happy New Year,

Just trying to work out what to do with the timeout notifications 
from RTPProxy when the RTPProxy is used by more than one Opensips proxy.


RTPProxy manual says:

/-n timeout_socket /

/This parameter configures the optional timeout notification
socket. The socket should be created by another application,
preferably before starting rtpproxy. For those sessions   where
the timeout mechanism is enabled, notifications are sent on this
socket if the session times out./

/There is no default value, notifications are not sent and not
permitted unless a value is specified explicitly. /

So if I have a primary and a backup SIP proxy, how do I get timeout 
notifications to both?


I have tried specifying one of the SIP proxies in the -n parameter 
and it works ok, but if I then put a call through the other SIP 
proxy , the RTPProxy logs:
Jan 13 13:00:13 rtpproxy1 rtpproxy[474]: 
ERR:1_395279493@192.168.6.48:rtpp_command_ul_handle: invalid socket 
name 81.xx.xxx.250:19991


I ask the question here because I know some of you do development 
work on both Opensips and RTPProxy.


Kind regards,

Adrian Fretwell
Nottinghamshire
UK.



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Re: [OpenSIPS-Users] RTPProxy Timeout socket more than one SIP proxy

[Adrian Fretwell]

Razvan,

Thankyou for clarifying this, I can work around it either with VIP or by 
restarting RTP Proxy with a different -n value.


Kind regards,

Adrian.

On 15/01/18 09:02, Răzvan Crainea wrote:

Hi, Adrian!

Unfortunately RTPProxy can send timeout notifications only to one 
timeout_socket - the one specified in the -n parameter. We did discuss 
with Maxim a while back the ability to support different 
timeout_sockets to different opensips instances, but that discussion 
didn't materialize yet.


However, for your setup, I guess you are already using a VIP(Virtual 
IP) that you share between the active and backup instance. An idea is 
to use the same VIP as the -n parameter - that way you will always 
send the timeout notification to the active OpenSIPS instance.


Let me know if that works for you.

Best regards,
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com
On 01/13/2018 03:26 PM, Adrian Fretwell wrote:

Hello & Happy New Year,

Just trying to work out what to do with the timeout notifications 
from RTPProxy when the RTPProxy is used by more than one Opensips proxy.


RTPProxy manual says:

/-n timeout_socket /

/This parameter configures the optional timeout notification
socket. The socket should be created by another application,
preferably before starting rtpproxy. For those sessions   where
the timeout mechanism is enabled, notifications are sent on this
socket if the session times out./

/There is no default value, notifications are not sent and not
permitted unless a value is specified explicitly. /

So if I have a primary and a backup SIP proxy, how do I get timeout 
notifications to both?


I have tried specifying one of the SIP proxies in the -n parameter 
and it works ok, but if I then put a call through the other SIP proxy 
, the RTPProxy logs:
Jan 13 13:00:13 rtpproxy1 rtpproxy[474]: 
ERR:1_395279493@192.168.6.48:rtpp_command_ul_handle: invalid socket 
name 81.xx.xxx.250:19991


I ask the question here because I know some of you do development 
work on both Opensips and RTPProxy.


Kind regards,

Adrian Fretwell
Nottinghamshire
UK.



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Re: [OpenSIPS-Users] RTPProxy Timeout socket more than one SIP proxy

[Răzvan Crainea]

Hi, Adrian!

Unfortunately RTPProxy can send timeout notifications only to one 
timeout_socket - the one specified in the -n parameter. We did discuss 
with Maxim a while back the ability to support different timeout_sockets 
to different opensips instances, but that discussion didn't materialize yet.


However, for your setup, I guess you are already using a VIP(Virtual IP) 
that you share between the active and backup instance. An idea is to use 
the same VIP as the -n parameter - that way you will always send the 
timeout notification to the active OpenSIPS instance.


Let me know if that works for you.

Best regards,

Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com

On 01/13/2018 03:26 PM, Adrian Fretwell wrote:

Hello & Happy New Year,

Just trying to work out what to do with the timeout notifications from 
RTPProxy when the RTPProxy is used by more than one Opensips proxy.


RTPProxy manual says:

/-n timeout_socket /

/This parameter configures the optional timeout notification
socket. The socket should be created by another application,
preferably before starting rtpproxy. For those sessions   where
the timeout mechanism is enabled, notifications are sent on this
socket if the session times out./

/There is no default value, notifications are not sent and not
permitted unless a value is specified explicitly. /

So if I have a primary and a backup SIP proxy, how do I get timeout 
notifications to both?


I have tried specifying one of the SIP proxies in the -n parameter and 
it works ok, but if I then put a call through the other SIP proxy , 
the RTPProxy logs:
Jan 13 13:00:13 rtpproxy1 rtpproxy[474]: 
ERR:1_395279493@192.168.6.48:rtpp_command_ul_handle: invalid socket 
name 81.xx.xxx.250:19991


I ask the question here because I know some of you do development work 
on both Opensips and RTPProxy.


Kind regards,

Adrian Fretwell
Nottinghamshire
UK.



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[OpenSIPS-Users] RTPProxy Timeout socket more than one SIP proxy

[Adrian Fretwell]

Hello & Happy New Year,

Just trying to work out what to do with the timeout notifications from 
RTPProxy when the RTPProxy is used by more than one Opensips proxy.


RTPProxy manual says:

   /-n timeout_socket /

   /This parameter configures the optional timeout notification socket.
   The socket should be created by another application, preferably
   before starting rtpproxy. For those sessions   where the timeout
   mechanism is enabled, notifications are sent on this socket if the
   session times out./

   /There is no default value, notifications are not sent and not
   permitted unless a value is specified explicitly. /

So if I have a primary and a backup SIP proxy, how do I get timeout 
notifications to both?


I have tried specifying one of the SIP proxies in the -n parameter and 
it works ok, but if I then put a call through the other SIP proxy , the 
RTPProxy logs:
Jan 13 13:00:13 rtpproxy1 rtpproxy[474]: 
ERR:1_395279493@192.168.6.48:rtpp_command_ul_handle: invalid socket name 
81.xx.xxx.250:19991


I ask the question here because I know some of you do development work 
on both Opensips and RTPProxy.


Kind regards,

Adrian Fretwell
Nottinghamshire
UK.

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Re: [OpenSIPS-Users] rtpproxy socket

[Denis via Users]

Hello, yes sure. 1. On initial INVITE store rtpproxy socket to any dialog value (store_dlg_value)2. In dialplan store rtpproxy socket (as match_exp) with attrs field, in which sock_id of rtpproxy has been inserted.3. Insert additional records to rtpproxy_sockets table with individual sock_id (correspond to attrs value in 2) ) for each rtpproxy instance.4. On re_INVITE fetch dialog value of rtpproxy socket5. Using dp_translate with rtpproxy socket value (as match_exp) find out sock_id of the rtpproxy (see 2) and attrs field)6. Finally, use rtpproxy_offer() with sock_id which you got in 5). -- С уважением, Денис.Best regards, Denis 12.09.2017, 10:57, "Serge S. Yuriev" :Hi Would you mind to share your experience with us?Thanx.--Wbr, Serge via mobile12.09.2017, 10:20, "Denis via Users" :Hello! I found solution myself. The question is closed. -- С уважением, Денис.Best regards, Denis 08.09.2017, 14:51, "Denis via Users" :Hello! Opensips 2.3 One interesting question about rtpproxy using. I can use multiple rtpproxy through one set. During each call Opensips chooses certain socket inside the certain set of proxies.Till this time everything clear.But during existing session SIP UA (caller or callee never mind) can initiate some re-Invite messages for some reason.And these messages should "marked" by rtpproxy too. And now the question, is there any way, during re-Invite processing, to tell Opensips to choose certain rtpproxy socket which has been chosen at the beginning of the call? Thank you for any help.  -- С уважением, Денис.Best regards, Denis   ,___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users,___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users___
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Re: [OpenSIPS-Users] rtpproxy socket

[Serge S . Yuriev]

HiWould you mind to share your experience with us?Thanx.-- Wbr, Serge via mobile12.09.2017, 10:20, "Denis via Users" :Hello! I found solution myself. The question is closed. -- С уважением, Денис.Best regards, Denis 08.09.2017, 14:51, "Denis via Users" :Hello! Opensips 2.3 One interesting question about rtpproxy using. I can use multiple rtpproxy through one set. During each call Opensips chooses certain socket inside the certain set of proxies.Till this time everything clear.But during existing session SIP UA (caller or callee never mind) can initiate some re-Invite messages for some reason.And these messages should "marked" by rtpproxy too. And now the question, is there any way, during re-Invite processing, to tell Opensips to choose certain rtpproxy socket which has been chosen at the beginning of the call? Thank you for any help.  -- С уважением, Денис.Best regards, Denis   ,___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users___
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Re: [OpenSIPS-Users] rtpproxy socket

[Denis via Users]

Hello! I found solution myself. The question is closed. -- С уважением, Денис.Best regards, Denis 08.09.2017, 14:51, "Denis via Users" :Hello! Opensips 2.3 One interesting question about rtpproxy using. I can use multiple rtpproxy through one set. During each call Opensips chooses certain socket inside the certain set of proxies.Till this time everything clear.But during existing session SIP UA (caller or callee never mind) can initiate some re-Invite messages for some reason.And these messages should "marked" by rtpproxy too. And now the question, is there any way, during re-Invite processing, to tell Opensips to choose certain rtpproxy socket which has been chosen at the beginning of the call? Thank you for any help.  -- С уважением, Денис.Best regards, Denis   ,___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users___
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[OpenSIPS-Users] rtpproxy socket

[Denis via Users]

Hello! Opensips 2.3 One interesting question about rtpproxy using. I can use multiple rtpproxy through one set. During each call Opensips chooses certain socket inside the certain set of proxies.Till this time everything clear.But during existing session SIP UA (caller or callee never mind) can initiate some re-Invite messages for some reason.And these messages should "marked" by rtpproxy too. And now the question, is there any way, during re-Invite processing, to tell Opensips to choose certain rtpproxy socket which has been chosen at the beginning of the call? Thank you for any help.  -- С уважением, Денис.Best regards, Denis   

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Re: [OpenSIPS-Users] rtpproxy not relaying data

[Waldoalvarez via Users]

Hi Răzvan Crainea:

I was sending an U command instead of an L command. This is what I interpreted 
in the documentation. Is working great now. Thanks a lot! This was worrying me 
a lot. I've being dealing with this RTP thing since long. Thought I had to 
implement handling that protocol. Thanks a lot!



Sent with [ProtonMail](https://protonmail.com) Secure Email.


 Original Message 
Subject: Re: [OpenSIPS-Users] rtpproxy not relaying data
Local Time: 22 de febrero de 2017 6:14 AM
UTC Time: 22 de febrero de 2017 10:14
From: raz...@opensips.org
To: users@lists.opensips.org

Hi, Waldo!

I only see the command for Update (initial request), I don't see the command 
for Lookup(200 OK).
Moreover, are you sure RTP traffic gets to the rtpproxy machine? Because 
RTPProxy statistics doesn't see any packets getting to the server. Can you 
double check if the firewall allows the traffic?

Best regards,

Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com

On 02/22/2017 11:44 AM, Waldoalvarez via Users wrote:

Hi:


I am trying to use rtpproxy with my SIP proxy. It starts a session and both SIP 
clients send data to rtpproxy port as I see on wireshark after I modify the SDP 
part. I see data comming to rtpproxy but no data comes out of it. What could be 
going on wrong? Configuratoin issue? I send here a log of rtpproxy.

halplus@halplus-VirtualBox ~ $ rtpproxy -f -s unix:/var/run/rtp/rtpproxy.sock 
-d DBUG:LOG_LOCAL3
INFO:main: rtpproxy started, pid 11266
ERR:main: can't open pidfile for writing: Permission denied
DBUG:handle_command: received command "U rWiXCizLxX 192.168.200.146 7076 
SA6YGw3uI"
INFO:handle_command: new session rWiXCizLxX, tag SA6YGw3uI requested, type 
strong
INFO:handle_command: new session on a port 37110 created, tag SA6YGw3uI
INFO:handle_command: pre-filling caller's address with 192.168.200.146:7076
DBUG:doreply: sending reply "37110
"
DBUG:handle_command: received command "U rWiXCizLxX 192.168.200.140 7078 
SA6YGw3uI"
INFO:handle_command: adding strong flag to existing session, new=1/0/0
INFO:handle_command: lookup on ports 37110/0, session timer restarted
INFO:handle_command: pre-filling caller's address with 192.168.200.140:7078
DBUG:doreply: sending reply "37110
"
INFO:process_rtp: session timeout
INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 
0 dropped
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 
0 dropped
INFO:remove_session: session on ports 37110/0 is cleaned up

Any help is very much welcome.

Regards
Waldo



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Re: [OpenSIPS-Users] rtpproxy not relaying data

[Răzvan Crainea]

Hi, Waldo!

I only see the command for Update (initial request), I don't see the 
command for Lookup(200 OK).
Moreover, are you sure RTP traffic gets to the rtpproxy machine? Because 
RTPProxy statistics doesn't see any packets getting to the server. Can 
you double check if the firewall allows the traffic?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 02/22/2017 11:44 AM, Waldoalvarez via Users wrote:


Hi:

I am trying to use rtpproxy with my SIP proxy. It starts a session and 
both SIP clients send data to rtpproxy port as I see on wireshark 
after I modify the SDP part. I see data comming to rtpproxy but no 
data comes out of it. What could be going on wrong? Configuratoin 
issue? I send here a log of rtpproxy.


halplus@halplus-VirtualBox ~ $ rtpproxy -f -s 
unix:/var/run/rtp/rtpproxy.sock -d DBUG:LOG_LOCAL3

INFO:main: rtpproxy started, pid 11266
ERR:main: can't open pidfile for writing: Permission denied
DBUG:handle_command: received command "U rWiXCizLxX 192.168.200.146 
7076 SA6YGw3uI"
INFO:handle_command: new session rWiXCizLxX, tag SA6YGw3uI requested, 
type strong

INFO:handle_command: new session on a port 37110 created, tag SA6YGw3uI
INFO:handle_command: pre-filling caller's address with 
192.168.200.146:7076

DBUG:doreply: sending reply "37110
"
DBUG:handle_command: received command "U rWiXCizLxX 192.168.200.140 
7078 SA6YGw3uI"

INFO:handle_command: adding strong flag to existing session, new=1/0/0
INFO:handle_command: lookup on ports 37110/0, session timer restarted
INFO:handle_command: pre-filling caller's address with 
192.168.200.140:7078

DBUG:doreply: sending reply "37110
"
INFO:process_rtp: session timeout
INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 
relayed, 0 dropped
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 
relayed, 0 dropped

INFO:remove_session: session on ports 37110/0 is cleaned up

Any help is very much welcome.

Regards
Waldo



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[OpenSIPS-Users] rtpproxy not relaying data

[Waldoalvarez via Users]

Hi:


I am trying to use rtpproxy with my SIP proxy. It starts a session and both SIP 
clients send data to rtpproxy port as I see on wireshark after I modify the SDP 
part. I see data comming to rtpproxy but no data comes out of it. What could be 
going on wrong? Configuratoin issue? I send here a log of rtpproxy.



halplus@halplus-VirtualBox ~ $ rtpproxy -f -s unix:/var/run/rtp/rtpproxy.sock 
-d DBUG:LOG_LOCAL3
INFO:main: rtpproxy started, pid 11266
ERR:main: can't open pidfile for writing: Permission denied
DBUG:handle_command: received command "U rWiXCizLxX 192.168.200.146 7076 
SA6YGw3uI"
INFO:handle_command: new session rWiXCizLxX, tag SA6YGw3uI requested, type 
strong
INFO:handle_command: new session on a port 37110 created, tag SA6YGw3uI
INFO:handle_command: pre-filling caller's address with 192.168.200.146:7076
DBUG:doreply: sending reply "37110
"
DBUG:handle_command: received command "U rWiXCizLxX 192.168.200.140 7078 
SA6YGw3uI"
INFO:handle_command: adding strong flag to existing session, new=1/0/0
INFO:handle_command: lookup on ports 37110/0, session timer restarted
INFO:handle_command: pre-filling caller's address with 192.168.200.140:7078
DBUG:doreply: sending reply "37110
"
INFO:process_rtp: session timeout
INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 
0 dropped
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 
0 dropped
INFO:remove_session: session on ports 37110/0 is cleaned up

Any help is very much welcome.

Regards
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Re: [OpenSIPS-Users] rtpproxy and record calls

[Denis via Users]

Hello, Razvan! 1) I found what the problem was. The question is closed.2) To your notice. -- С уважением, Денис.Best regards, Denis 09.02.2017, 15:38, "Denis via Users" :Hello Razvan! I am sorry for long answer, only now could return to these questions. So 1) I run rtpproxy_start_recording() for both INVITE and 200 OK.The some content of the opensips.cfg"...modparam("rtpproxy", "rtpproxy_sock", "1 == udp::221") route[1]if (is_method("INVITE")&&!has_totag()) {   
if (isflagset(10)) {
  if ($avp(3000)==1)  xlog("L_INFO", "Route1:$rm was received and call_record detected(IPS=$si, SP=$sp,  IPD=$rd, RP=$Rp,  FROMTAG=$ft, TOTAG=$tt)");
   rtpproxy_start_recording("1");
 }
}. onreply_route[1]if (isflagset(10) && $avp(1003)==1) {
  rtpproxy_start_recording("1");  
}"In syslog of the Opensips instance i can see " Route1:INVITE was received and call_record detected(IPS=, SP=5068,  IPD=, RP=5065,  FROMTAG=5reZ564rtpHBg, TOTAG=)"In syslog of the rtpproxy instance i can see "INFO:handle_copy:DLGCH_KklSW2IoYjFiRlxYZ2RhYXxFSFdlK2R+f0JXXmt5MWspSVEM: starting recording RTP session on port 38598"but only when 200 OK has been received. I usertpproxy_engage("conrf",,"1",);for make an rtp proxy. 2) I understand that "/" is a valid character, but, to speak the truth, did not invite it in any call-id before. Only when started using topo_hiding with Opensips. May i ask to make some patch to remove "/" from callid generation? Thank you.-- С уважением, Денис.Best regards, Denis 04.01.2017, 18:55, "Răzvan Crainea" :Hi, Denis!Regarding 1, did you try to run rtpproxy_start_recording() for both INVITE and 200 OK?Regarding 2, '/' is a valid character in Call-id. Therefore the problem is at the RTPProxy side - before writing the CDR they should escape (or transform somehow) the '/' character in something else.Best regards,Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.comOn 01/04/2017 05:24 PM, Денис Путято via Users wrote:Hello! Is there any information about the problem? Thank you.--С уважением,Путято ДенисBest regards, Denis14:47, 27 декабря 2016 г., Denis via Users :Hello! I try to use rtpproxy for call recording and have two problems 1) rtpproxy records only one way of the call (from callee to caller). For starting rtp proxy i use rtpproxy_engage("conrf",,"1",) function. 2) i am using top_hiding with "C" flags (which should change callid) and i noticed, that sometime the function generates callid with such form (for example)DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM. I want to pay attention to "/" character. As i understand, because of this character i got such error from rtpproxyERR:DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM:rtpp_record_open: can't open file /mnt/callrecords//DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM=cU7ZHtge63paN.a.rtp for writing: No such file or directory (2) rtpproxy: 2.2.alpha.20160822Opensips: Server:: OpenSIPS (2.2.2 (x86_64/linux)) Thank you for any help. -- С уважением, Денис.Best regards, Denis   ___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___
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Re: [OpenSIPS-Users] rtpproxy and record calls

[Denis via Users]

Hello Razvan! I am sorry for long answer, only now could return to these questions. So 1) I run rtpproxy_start_recording() for both INVITE and 200 OK.The some content of the opensips.cfg"...modparam("rtpproxy", "rtpproxy_sock", "1 == udp::221") route[1]if (is_method("INVITE")&&!has_totag()) {   
if (isflagset(10)) {
  if ($avp(3000)==1)  xlog("L_INFO", "Route1:$rm was received and call_record detected(IPS=$si, SP=$sp,  IPD=$rd, RP=$Rp,  FROMTAG=$ft, TOTAG=$tt)");
   rtpproxy_start_recording("1");
 }
}. onreply_route[1]if (isflagset(10) && $avp(1003)==1) {
  rtpproxy_start_recording("1");  
}"In syslog of the Opensips instance i can see " Route1:INVITE was received and call_record detected(IPS=, SP=5068,  IPD=, RP=5065,  FROMTAG=5reZ564rtpHBg, TOTAG=)"In syslog of the rtpproxy instance i can see "INFO:handle_copy:DLGCH_KklSW2IoYjFiRlxYZ2RhYXxFSFdlK2R+f0JXXmt5MWspSVEM: starting recording RTP session on port 38598"but only when 200 OK has been received. I usertpproxy_engage("conrf",,"1",);for make an rtp proxy. 2) I understand that "/" is a valid character, but, to speak the truth, did not invite it in any call-id before. Only when started using topo_hiding with Opensips. May i ask to make some patch to remove "/" from callid generation? Thank you.-- С уважением, Денис.Best regards, Denis 04.01.2017, 18:55, "Răzvan Crainea" :Hi, Denis!Regarding 1, did you try to run rtpproxy_start_recording() for both INVITE and 200 OK?Regarding 2, '/' is a valid character in Call-id. Therefore the problem is at the RTPProxy side - before writing the CDR they should escape (or transform somehow) the '/' character in something else.Best regards,Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.comOn 01/04/2017 05:24 PM, Денис Путято via Users wrote:Hello! Is there any information about the problem? Thank you.--С уважением,Путято ДенисBest regards, Denis14:47, 27 декабря 2016 г., Denis via Users :Hello! I try to use rtpproxy for call recording and have two problems 1) rtpproxy records only one way of the call (from callee to caller). For starting rtp proxy i use rtpproxy_engage("conrf",,"1",) function. 2) i am using top_hiding with "C" flags (which should change callid) and i noticed, that sometime the function generates callid with such form (for example)DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM. I want to pay attention to "/" character. As i understand, because of this character i got such error from rtpproxyERR:DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM:rtpp_record_open: can't open file /mnt/callrecords//DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM=cU7ZHtge63paN.a.rtp for writing: No such file or directory (2) rtpproxy: 2.2.alpha.20160822Opensips: Server:: OpenSIPS (2.2.2 (x86_64/linux)) Thank you for any help. -- С уважением, Денис.Best regards, Denis   ___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___
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Re: [OpenSIPS-Users] rtpproxy and SRTP

[John Quick]

Hi Sasmita,

Thanks for the information.
My main objective will be the same basic scenario that you described. That is, 
simple bridging between two end points.

I was concerned that the rtpproxy product includes support for options like
 - recording of audio
 - payload re-framing

For these options to work, it must require deeper access to the encrypted data 
than is needed for simple "blind" proxying of RTP packets.
In the documentation I could not see anything about certificates. So, like you, 
I am curious to know the answer.
However, as I do not need these options I hope there will be no problem. I am 
setting up a test environment at the moment.

John Quick
Smartvox Limited


From: Sasmita Panda [mailto:spa...@3clogic.com] 
Sent: 07 February 2017 06:54
To: john.qu...@smartvox.co.uk; OpenSIPS users mailling list 

Subject: Re: [OpenSIPS-Users] rtpproxy and SRTP

Hi , 

I have tested 1 scenario . If there is two end point which support 
encrypted media (SRTP) and there is rtpproxy between them . Then rtpproxy works 
as usual . It update the C line Ip in the SDP and forwards the request and 
response .  This is what I tested . And its working . 
  
   I don't know which prospective you are asking rtpproxy supports SRTP or 
not . If you mean anything different then I am also curious to know .
 


Thanks & Regards
Sasmita Panda
Network Testing and Software Engineer
3CLogic , ph:07827611765

On Mon, Feb 6, 2017 at 7:13 PM, John Quick <mailto:john.qu...@smartvox.co.uk> 
wrote:
Please, does anyone know if rtpproxy works with SRTP?

John Quick
Smartvox Limited



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Re: [OpenSIPS-Users] rtpproxy and SRTP

[Sasmita Panda]

Hi ,

I have tested 1 scenario . If there is two end point which support
encrypted media (SRTP) and there is rtpproxy between them . Then rtpproxy
works as usual . It update the C line Ip in the SDP and forwards the
request and response .  This is what I tested . And its working .

   I don't know which prospective you are asking rtpproxy supports SRTP
or not . If you mean anything different then I am also curious to know .


*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Mon, Feb 6, 2017 at 7:13 PM, John Quick 
wrote:

> Please, does anyone know if rtpproxy works with SRTP?
>
> John Quick
> Smartvox Limited
>
>
>
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[OpenSIPS-Users] rtpproxy and SRTP

[John Quick]

Please, does anyone know if rtpproxy works with SRTP?

John Quick
Smartvox Limited



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Re: [OpenSIPS-Users] Rtpproxy bug

[Bogdan-Andrei Iancu]

Hi Robert,

It looks like a bug to me, but you have to check and report this to 
someone having commit rights on the rtpproxy repository.


Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 02/04/2017 01:59 AM, Robert Dyck wrote:

I am reporting here because I don't know how to leave a bug report with Sippy
Software.

When I installed opensips and configured it use use rtpproxy and the unix
socket, opensips would not start. It turned out to be an issue with
permissions. It was simple to change to the loopback instead but I was curious
and wanted to get to the root of the problem.

I experimented with the -w command line option of rtpproxy. Combined with the
-u option this is supposed to allow one to set the permissions for the socket.
The default mode for the socket is 755 and I tried setting it to 775. Setting
-w 775 resulted in something very different and further changes did not yield
predictable results.

I looked into the code and it uses the atoi function to convert the mode
string into an integer rather than an octal number. Working backwards from 775
I get 0x1FD and 509 decimal. If I set the mode to 509 I get the desired
result.

Version info
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the update/lookup command
Extension 20081224: Support for session timeout notifications
Extension 20090810: Support for automatic bridging
Extension 20140323: Support for tracking/reporting load
Extension 20140617: Support for anchoring session connect time
Extension 20141004: Support for extendable performance counters

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[OpenSIPS-Users] Rtpproxy bug

[Robert Dyck]

I am reporting here because I don't know how to leave a bug report with Sippy 
Software.

When I installed opensips and configured it use use rtpproxy and the unix 
socket, opensips would not start. It turned out to be an issue with 
permissions. It was simple to change to the loopback instead but I was curious 
and wanted to get to the root of the problem.

I experimented with the -w command line option of rtpproxy. Combined with the 
-u option this is supposed to allow one to set the permissions for the socket. 
The default mode for the socket is 755 and I tried setting it to 775. Setting 
-w 775 resulted in something very different and further changes did not yield 
predictable results.

I looked into the code and it uses the atoi function to convert the mode 
string into an integer rather than an octal number. Working backwards from 775 
I get 0x1FD and 509 decimal. If I set the mode to 509 I get the desired 
result.

Version info
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the update/lookup command
Extension 20081224: Support for session timeout notifications
Extension 20090810: Support for automatic bridging
Extension 20140323: Support for tracking/reporting load
Extension 20140617: Support for anchoring session connect time
Extension 20141004: Support for extendable performance counters

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Re: [OpenSIPS-Users] rtpproxy and record calls

[Răzvan Crainea]

Hi, Denis!

Regarding 1, did you try to run rtpproxy_start_recording() for both 
INVITE and 200 OK?


Regarding 2, '/' is a valid character in Call-id. Therefore the problem 
is at the RTPProxy side - before writing the CDR they should escape (or 
transform somehow) the '/' character in something else.


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 01/04/2017 05:24 PM, Денис Путято via Users wrote:

Hello!

Is there any information about the problem?

Thank you.

--
С уважением,
Путято Денис
Best regards, Denis

14:47, 27 декабря 2016 г., Denis via Users :

Hello!
I try to use rtpproxy for call recording and have two problems
1) rtpproxy records only one way of the call (from callee to
caller). For starting rtp proxy i use
rtpproxy_engage("conrf",,"1",) function.
2) i am using top_hiding with "C" flags (which should change
callid) and i noticed, that sometime the function generates callid
with such form (for example)
DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM.
I want to pay attention to "/" character. As i understand, because
of this character i got such error from rtpproxy
ERR:DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM:rtpp_record_open:
can't open file

/mnt/callrecords//DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM=cU7ZHtge63paN.a.rtp
for writing: No such file or directory (2)
rtpproxy: 2.2.alpha.20160822
Opensips: Server:: OpenSIPS (2.2.2 (x86_64/linux))
Thank you for any help.
-- 
С уважением, Денис.

Best regards, Denis

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Re: [OpenSIPS-Users] rtpproxy and record calls

[Денис Путято via Users]

Hello!Is there any information about the problem?Thank you.-- С уважением,Путято ДенисBest regards, Denis14:47, 27 декабря 2016 г., Denis via Users :Hello! I try to use rtpproxy for call recording and have two problems 1) rtpproxy records only one way of the call (from callee to caller). For starting rtp proxy i use rtpproxy_engage("conrf",,"1",) function. 2) i am using top_hiding with "C" flags (which should change callid) and i noticed, that sometime the function generates callid with such form (for example)DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM. I want to pay attention to "/" character. As i understand, because of this character i got such error from rtpproxyERR:DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM:rtpp_record_open: can't open file /mnt/callrecords//DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM=cU7ZHtge63paN.a.rtp for writing: No such file or directory (2) rtpproxy: 2.2.alpha.20160822Opensips: Server:: OpenSIPS (2.2.2 (x86_64/linux)) Thank you for any help. -- С уважением, Денис.Best regards, Denis   ___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
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[OpenSIPS-Users] rtpproxy and record calls

[Denis via Users]

Hello! I try to use rtpproxy for call recording and have two problems 1) rtpproxy records only one way of the call (from callee to caller). For starting rtp proxy i use rtpproxy_engage("conrf",,"1",) function. 2) i am using top_hiding with "C" flags (which should change callid) and i noticed, that sometime the function generates callid with such form (for example)DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM. I want to pay attention to "/" character. As i understand, because of this character i got such error from rtpproxyERR:DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM:rtpp_record_open: can't open file /mnt/callrecords//DLGCH_fRUAW2B/M2NiSV1WYWRhYX9ASFowK2J+f0JXXmt5MWspSVEM=cU7ZHtge63paN.a.rtp for writing: No such file or directory (2) rtpproxy: 2.2.alpha.20160822Opensips: Server:: OpenSIPS (2.2.2 (x86_64/linux)) Thank you for any help. -- С уважением, Денис.Best regards, Denis   ___
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Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

[Denis via Users]

Hello, Flavio! Thank you very much for your help! I made some test and it worked.But, in additional, i want to ask you about g729 codec. In rtpproxy documentation says that beginning from 2.0 g729 codec supported.In the dictionary of extractaudo i see some g729* files. Is there necessary to compile bcg729, or it needs exactly for extractaudio? Thank you.  -- С уважением, Денис.Best regards, Denis   27.12.2016, 11:27, "Flavio Goncalves" :Hi,  Yes you can extract audio from rtpproxy. The extractaudio utility is very handy and you can compile with G.729 from the linphone project bcg729. It is very easy to use, simply  use the utility followed by the name of the recording without any extension. Check the source code for the other options. Running without parameters gives you the command syntax. The only tip, is save the rtp in PCAP mode and  don't use the file extension when calling the command. It mixes both channels together automagically.  Best regards,  Flavio E. Goncalves   2016-12-24 11:07 GMT-02:00 Денис Путято via Users :I read about it, but if i need extract g729?--С уважением,Путято ДенисBest regards, Denis18:08, 23 декабря 2016 г., Ovidiu Sas : Here are the steps to extract the audio:https://voipembedded.wordpress.com/2011/11/15/extracting-audio-from-calls-recorded-with-rtpproxy/ Regards, Ovidiu Sas  On Dec 23, 2016 09:52, "Denis via Users"  wrote:Hello, Bogdan! I mean haw can i extract audio from .rtp file, which was recorded by rtpproxy.In documentation to rtpproxy said, that extractaudio utility can do that. But how do that, i cannot find. -- С уважением, Денис.Best regards, Denis   23.12.2016, 17:49, "Bogdan-Andrei Iancu" :Hi Denis,You mean to record the RTP going through rtpproxy ?Regards,Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.comOn 23.12.2016 12:22, Denis wrote:Hello! I want to ask about extractaudio. Is there any manual for it? How can i use it?Has anybody operation experience of this soft? Thank you. -- С уважением, Денис.Best regards, Denis___
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Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

[Flavio Goncalves]

Hi,

Yes you can extract audio from rtpproxy. The extractaudio utility is very
handy and you can compile with G.729 from the linphone project bcg729. It
is very easy to use, simply  use the utility followed by the name of the
recording without any extension. Check the source code for the other
options. Running without parameters gives you the command syntax. The only
tip, is save the rtp in PCAP mode and  don't use the file extension when
calling the command. It mixes both channels together automagically.

Best regards,

Flavio E. Goncalves



2016-12-24 11:07 GMT-02:00 Денис Путято via Users 
:

> I read about it, but if i need extract g729?
>
> --
> С уважением,
> Путято Денис
> Best regards, Denis
>
> 18:08, 23 декабря 2016 г., Ovidiu Sas :
>
> Here are the steps to extract the audio:
> https://voipembedded.wordpress.com/2011/11/15/extracting-audio-from-calls-
> recorded-with-rtpproxy/
>
> Regards,
> Ovidiu Sas
>
>
> On Dec 23, 2016 09:52, "Denis via Users"  wrote:
>
> Hello, Bogdan!
>
> I mean haw can i extract audio from .rtp file, which was recorded by
> rtpproxy.
> In documentation to rtpproxy said, that extractaudio utility can do that.
> But how do that, i cannot find.
>
> --
> С уважением, Денис.
> Best regards, Denis
>
>
>
> 23.12.2016, 17:49, "Bogdan-Andrei Iancu" :
>
> Hi Denis,
>
> You mean to record the RTP going through rtpproxy ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 23.12.2016 12:22, Denis wrote:
>
> Hello!
>
> I want to ask about extractaudio. Is there any manual for it? How can i
> use it?
> Has anybody operation experience of this soft?
>
> Thank you.
>
> --
> С уважением, Денис.
> Best regards, Denis
>
>
>
>
>
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Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

[Денис Путято via Users]

I read about it, but if i need extract g729?-- С уважением,Путято ДенисBest regards, Denis18:08, 23 декабря 2016 г., Ovidiu Sas :Here are the steps to extract the audio:https://voipembedded.wordpress.com/2011/11/15/extracting-audio-from-calls-recorded-with-rtpproxy/Regards, Ovidiu SasOn Dec 23, 2016 09:52, "Denis via Users"  wrote:Hello, Bogdan! I mean haw can i extract audio from .rtp file, which was recorded by rtpproxy.In documentation to rtpproxy said, that extractaudio utility can do that. But how do that, i cannot find. -- С уважением, Денис.Best regards, Denis   23.12.2016, 17:49, "Bogdan-Andrei Iancu" :Hi Denis,You mean to record the RTP going through rtpproxy ?Regards,Bogdan-Andrei IancuOpenSIPS Founder and Developerhttp://www.opensips-solutions.comOn 23.12.2016 12:22, Denis wrote:Hello! I want to ask about extractaudio. Is there any manual for it? How can i use it?Has anybody operation experience of this soft? Thank you. -- С уважением, Денис.Best regards, Denis___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users___Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

[Ovidiu Sas]

Here are the steps to extract the audio:
https://voipembedded.wordpress.com/2011/11/15/extracting-audio-from-calls-recorded-with-rtpproxy/

Regards,
Ovidiu Sas


On Dec 23, 2016 09:52, "Denis via Users"  wrote:

Hello, Bogdan!

I mean haw can i extract audio from .rtp file, which was recorded by
rtpproxy.
In documentation to rtpproxy said, that extractaudio utility can do that.
But how do that, i cannot find.

-- 
С уважением, Денис.
Best regards, Denis



23.12.2016, 17:49, "Bogdan-Andrei Iancu" :

Hi Denis,

You mean to record the RTP going through rtpproxy ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com

On 23.12.2016 12:22, Denis wrote:

Hello!

I want to ask about extractaudio. Is there any manual for it? How can i use
it?
Has anybody operation experience of this soft?

Thank you.

-- 
С уважением, Денис.
Best regards, Denis





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Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

[Bogdan-Andrei Iancu]

Denis,

as I see, this is a tool provided by the rtpproxy project - you should 
dig and ask more on their side.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 23.12.2016 16:51, Denis wrote:

Hello, Bogdan!
I mean haw can i extract audio from .rtp file, which was recorded by 
rtpproxy.
In documentation to rtpproxy said, that extractaudio utility can do 
that. But how do that, i cannot find.

--
С уважением, Денис.
Best regards, Denis
23.12.2016, 17:49, "Bogdan-Andrei Iancu" :

Hi Denis,

You mean to record the RTP going through rtpproxy ?

Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com 
On 23.12.2016 12:22, Denis wrote:

Hello!
I want to ask about extractaudio. Is there any manual for it? How 
can i use it?

Has anybody operation experience of this soft?
Thank you.
--
С уважением, Денис.
Best regards, Denis
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Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

[Denis via Users]

Hello, Bogdan! I mean haw can i extract audio from .rtp file, which was recorded by rtpproxy.In documentation to rtpproxy said, that extractaudio utility can do that. But how do that, i cannot find. -- С уважением, Денис.Best regards, Denis   23.12.2016, 17:49, "Bogdan-Andrei Iancu" :Hi Denis,You mean to record the RTP going through rtpproxy ?Regards,Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.comOn 23.12.2016 12:22, Denis wrote:Hello! I want to ask about extractaudio. Is there any manual for it? How can i use it?Has anybody operation experience of this soft? Thank you. -- С уважением, Денис.Best regards, Denis___
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Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

[Bogdan-Andrei Iancu]

Hi Denis,

You mean to record the RTP going through rtpproxy ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 23.12.2016 12:22, Denis wrote:

Hello!
I want to ask about extractaudio. Is there any manual for it? How can 
i use it?

Has anybody operation experience of this soft?
Thank you.
--
С уважением, Денис.
Best regards, Denis


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[OpenSIPS-Users] rtpproxy 2.0 and extractaudio

[Denis]

Hello! I want to ask about extractaudio. Is there any manual for it? How can i use it?Has anybody operation experience of this soft? Thank you. -- С уважением, Денис.Best regards, Denis   

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Re: [OpenSIPS-Users] rtpproxy and timeout socket

[Răzvan Crainea]

Yes, It seems OpenSIPS is not compatible with rtpproxy 2 version for 
timeout notifications. That's because OpenSIPS always strips the tcp: 
from the beginning of the socket, while RTPProxy 2 always waits for this 
prefix.


Please open a bug/feature request on github for us to track this issue:
https://github.com/OpenSIPS/opensips/issues

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 12/21/2016 10:59 AM, Denis wrote:

Hello, Razvan!
2.2.alpha.20160822
--
С уважением, Денис.
Best regards, Denis
21.12.2016, 11:58, "Răzvan Crainea" :

Hi, Denis!

What version of rtpproxy are you using? There might be an 
incompatibility issue here.


Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com 
On 12/21/2016 07:37 AM, Denis wrote:

Hello!
I am using rtpproxy in my VoIP network and now i want to locate it 
on another server rather then Opensips instance.

Rtpproxy is started with such parameters:
/usr/local/rtpproxy2/bin/rtpproxy -u  -p  -l 
 -s udp:: -i -T 10 -n tcp::2229 -m 35000 -M 45000 
-r  -R -P -d INFO LOG_LOCAL5
The Proxy of rtp packet works fine but when i emulated stop transfer 
of rtp i see in log such string:

ERR::rtpp_command_ul_handle: invalid socket name :2229
What the problem is?
Thank you for any help.
--
С уважением, Денис.
Best regards, Denis
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Re: [OpenSIPS-Users] rtpproxy and timeout socket

[Denis]

Hello, Razvan! 2.2.alpha.20160822 -- С уважением, Денис.Best regards, Denis 21.12.2016, 11:58, "Răzvan Crainea" :Hi, Denis!What version of rtpproxy are you using? There might be an incompatibility issue here.Best regards,Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.comOn 12/21/2016 07:37 AM, Denis wrote:Hello! I am using rtpproxy in my VoIP network and now i want to locate it on another server rather then Opensips instance.Rtpproxy is started with such parameters:/usr/local/rtpproxy2/bin/rtpproxy -u  -p  -l  -s udp:: -i -T 10 -n tcp::2229 -m 35000 -M 45000 -r  -R -P -d INFO LOG_LOCAL5 The Proxy of rtp packet works fine but when i emulated stop transfer of rtp i see in log such string:ERR::rtpp_command_ul_handle: invalid socket name :2229 What the problem is? Thank you for any help. -- С уважением, Денис.Best regards, Denis___
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Re: [OpenSIPS-Users] rtpproxy and timeout socket

[Denis]

Hello, Razvan! 2.2.alpha.20160822 -- С уважением, Денис.Best regards, Denis 21.12.2016, 11:58, "Răzvan Crainea" :Hi, Denis!What version of rtpproxy are you using? There might be an incompatibility issue here.Best regards,Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.comOn 12/21/2016 07:37 AM, Denis wrote:Hello! I am using rtpproxy in my VoIP network and now i want to locate it on another server rather then Opensips instance.Rtpproxy is started with such parameters:/usr/local/rtpproxy2/bin/rtpproxy -u  -p  -l  -s udp:: -i -T 10 -n tcp::2229 -m 35000 -M 45000 -r  -R -P -d INFO LOG_LOCAL5 The Proxy of rtp packet works fine but when i emulated stop transfer of rtp i see in log such string:ERR::rtpp_command_ul_handle: invalid socket name :2229 What the problem is? Thank you for any help. -- С уважением, Денис.Best regards, Denis___
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Re: [OpenSIPS-Users] rtpproxy and timeout socket

[Răzvan Crainea]

Hi, Denis!

What version of rtpproxy are you using? There might be an 
incompatibility issue here.


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 12/21/2016 07:37 AM, Denis wrote:

Hello!
I am using rtpproxy in my VoIP network and now i want to locate it on 
another server rather then Opensips instance.

Rtpproxy is started with such parameters:
/usr/local/rtpproxy2/bin/rtpproxy -u  -p  -l 
 -s udp:: -i -T 10 -n tcp::2229 -m 35000 -M 45000 -r 
 -R -P -d INFO LOG_LOCAL5
The Proxy of rtp packet works fine but when i emulated stop transfer 
of rtp i see in log such string:

ERR::rtpp_command_ul_handle: invalid socket name :2229
What the problem is?
Thank you for any help.
--
С уважением, Денис.
Best regards, Denis


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[OpenSIPS-Users] rtpproxy and timeout socket

[Denis]

Hello! I am using rtpproxy in my VoIP network and now i want to locate it on another server rather then Opensips instance.Rtpproxy is started with such parameters:/usr/local/rtpproxy2/bin/rtpproxy -u  -p  -l  -s udp:: -i -T 10 -n tcp::2229 -m 35000 -M 45000 -r  -R -P -d INFO LOG_LOCAL5 The Proxy of rtp packet works fine but when i emulated stop transfer of rtp i see in log such string:ERR::rtpp_command_ul_handle: invalid socket name :2229 What the problem is? Thank you for any help. -- С уважением, Денис.Best regards, Denis   

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Re: [OpenSIPS-Users] rtpproxy compile

[Denis]

Just one additional question.Is there any documentation about extractaudio utility?I cannot find any README file about it in the package and http://www.rtpproxy.org/post/v2release/ just mention about it. -- С уважением, Денис.Best regards, Denis 20.12.2016, 11:15, "Răzvan Crainea" :Hi, Denis!Can you run 'ldconfig -p | grep sndfile'? Do you see the .so libraries in the output? If not, perhaps you should run 'ldconfig'. If this still does not work, you should manually add the library's directory in the library path[1].If you do get sndfile in the output, but it still does not work, perhaps the first configure commmand did not get the proper library, so you should re-run './configure'.PS: I also forwarded this question on the RTPProxy mailing list. If this does not work, perhaps somebody out there can help you out.[1] https://codeyarns.com/2014/01/14/how-to-add-library-directory-to-ldconfig-cache/Best regards,Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.comOn 12/20/2016 08:46 AM, Denis wrote:Hello! I want to use extractaudio utility to extract audio and write it to some .wav file.But, during compiling of this utility i get such error"extractaudio.c:325: undefined reference to `sf_open'extractaudio.c:389: undefined reference to `sf_write_short'extractaudio.c:400: undefined reference to `sf_close'extractaudio.c:389: undefined reference to `sf_write_short' collect2: error: ld returned 1 exit statusMakefile:427: recipe for target 'extractaudio' failedmake: *** [extractaudio] Error 1"libsndfile1 package (and dev) had been installed early. May be, somebody, has dealt with the problem? Thank you for any help. -- С уважением, Денис.Best regards, Denis___
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Re: [OpenSIPS-Users] rtpproxy compile

[Denis]

Hello, Razvan! Thank you very much for your help! You are right, i should make additional ./configure after libsndfile installed. Now everything work fine! Thank you.  -- С уважением, Денис.Best regards, Denis 20.12.2016, 11:15, "Răzvan Crainea" :Hi, Denis!Can you run 'ldconfig -p | grep sndfile'? Do you see the .so libraries in the output? If not, perhaps you should run 'ldconfig'. If this still does not work, you should manually add the library's directory in the library path[1].If you do get sndfile in the output, but it still does not work, perhaps the first configure commmand did not get the proper library, so you should re-run './configure'.PS: I also forwarded this question on the RTPProxy mailing list. If this does not work, perhaps somebody out there can help you out.[1] https://codeyarns.com/2014/01/14/how-to-add-library-directory-to-ldconfig-cache/Best regards,Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.comOn 12/20/2016 08:46 AM, Denis wrote:Hello! I want to use extractaudio utility to extract audio and write it to some .wav file.But, during compiling of this utility i get such error"extractaudio.c:325: undefined reference to `sf_open'extractaudio.c:389: undefined reference to `sf_write_short'extractaudio.c:400: undefined reference to `sf_close'extractaudio.c:389: undefined reference to `sf_write_short' collect2: error: ld returned 1 exit statusMakefile:427: recipe for target 'extractaudio' failedmake: *** [extractaudio] Error 1"libsndfile1 package (and dev) had been installed early. May be, somebody, has dealt with the problem? Thank you for any help. -- С уважением, Денис.Best regards, Denis___
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Re: [OpenSIPS-Users] rtpproxy compile

[Răzvan Crainea]

Hi, Denis!

Can you run 'ldconfig -p | grep sndfile'? Do you see the .so libraries 
in the output? If not, perhaps you should run 'ldconfig'. If this still 
does not work, you should manually add the library's directory in the 
library path[1].


If you do get sndfile in the output, but it still does not work, perhaps 
the first configure commmand did not get the proper library, so you 
should re-run './configure'.


PS: I also forwarded this question on the RTPProxy mailing list. If this 
does not work, perhaps somebody out there can help you out.


[1] 
https://codeyarns.com/2014/01/14/how-to-add-library-directory-to-ldconfig-cache/


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 12/20/2016 08:46 AM, Denis wrote:

Hello!
I want to use extractaudio utility to extract audio and write it to 
some .wav file.

But, during compiling of this utility i get such error
"
extractaudio.c:325: undefined reference to `sf_open'
extractaudio.c:389: undefined reference to `sf_write_short'
extractaudio.c:400: undefined reference to `sf_close'
extractaudio.c:389: undefined reference to `sf_write_short'
collect2: error: ld returned 1 exit status
Makefile:427: recipe for target 'extractaudio' failed
make: *** [extractaudio] Error 1
"
libsndfile1 package (and dev) had been installed early.
May be, somebody, has dealt with the problem?
Thank you for any help.
--
С уважением, Денис.
Best regards, Denis


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[OpenSIPS-Users] rtpproxy compile

[Denis]

Hello! I want to use extractaudio utility to extract audio and write it to some .wav file.But, during compiling of this utility i get such error"extractaudio.c:325: undefined reference to `sf_open'extractaudio.c:389: undefined reference to `sf_write_short'extractaudio.c:400: undefined reference to `sf_close'extractaudio.c:389: undefined reference to `sf_write_short' collect2: error: ld returned 1 exit statusMakefile:427: recipe for target 'extractaudio' failedmake: *** [extractaudio] Error 1"libsndfile1 package (and dev) had been installed early. May be, somebody, has dealt with the problem? Thank you for any help. -- С уважением, Денис.Best regards, Denis   

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Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

[xiaofeng]

Hi, Robert,

Does rtpproxy _autobridge work?

http://www.opensips.org/html/docs/modules/1.11.x/rtpproxy#id293590

Regards,
xiaofeng
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Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

[Răzvan Crainea]

Hi, Robert!

Yes, in cases where you don't need IPv6, use II for those requests.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/09/2016 07:12 PM, Robert Dyck wrote:

I should have described the scenario in more detail.

The rtproxy is in bridge mode because two addresses were specified. This was to
accommodate IPV4 - IPV6 interworking. However the rtpproxy is also to be used
for NAT traversal. This is not a bridge in the physical sense because there
only one interface. For NAT traversal the IPV6 address should be ignored. Am I
correct in thinking that one should use either II flags or EE flags depending on
the order of the addresses given to rtpproxy?

Thank you for taking the time for this.

On November 9, 2016 12:23:09 PM you wrote:

Hi, Robert!

Yes, the I and E parameters are mandatory, and they should describe how
the RTP will flow. For example if the flow is from IPv4 to IPv6, you
should use EI; if the flow is from IPv4 to IPv6, then you should use IE.
And so on, depending on the call flow.

Regarding the address parameter, that is used when you want to overwrite
the address indicated by RTPProxy. This is used mainly for setups where
RTPProxy is behind NAT and the address inidcated is the private one. You
should swap this IP with the public advertised one.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/08/2016 09:51 PM, Robert Dyck wrote:

Thank you

Assuming rtpproxy was started with IPV4 as the first address and IPV6 as
the second, then in the NAT scenario, are the II flags mandatory in
offer/answer?

Slightly off topic, what sort of scenario would require the address
parameter for offer/answer?

On November 8, 2016 09:57:30 AM Răzvan Crainea wrote:

Hi, Robert!

See my answers inline.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/08/2016 02:15 AM, Robert Dyck wrote:

I have some question regarding rtpproxy capabilities in relation to
IPV4-IPV6 interworking.

The articles I have read say that you need to assign an address from
each
address family to rtpproxy. They go on to say that rtpproxy will then be
in
bridged mode. Others define bridge mode as assigning two interfaces to
rtpproxy.

As long as you have RTPProxy listening on two IPs, you have it set in
bridge mode. It doesn't matther whether one of them is IPv6, or both are.


If the IPV4 and IPV6 addresses are on the same interface, is the
rtpproxy
indeed in bridged mode? Should one avoid the use of engage_rtpproxy?

Yes, as stated above, RTPProxy is in bridged mode and you should avoid
using engage_rtpproxy(). That's because the function can't know/decide
which interface is which and cannot map with the RTPProxy's one.


Assuming that IPV4- IPV6 interworking is actually possible using
opensips
and rtpproxy, does that mean that an instance of rtpproxy is not
available to enable NAT traversal - would NAT traversal require using
another instance of rtpproxy using a single IPV4 address?

No, you don't need an extra instance - a single instance will do both
bridging and nat traversal.


Furthermore is the multihome parameter relevant to IPV4-IPV6
interworking
if opensips only listens on one interface?

The multihome parameter is only relevant for OpenSIPS, it doesn't
influence RTPProxy's behavior at all.

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Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

[Răzvan Crainea]

Hi, Robert!

Yes, the I and E parameters are mandatory, and they should describe how 
the RTP will flow. For example if the flow is from IPv4 to IPv6, you 
should use EI; if the flow is from IPv4 to IPv6, then you should use IE. 
And so on, depending on the call flow.


Regarding the address parameter, that is used when you want to overwrite 
the address indicated by RTPProxy. This is used mainly for setups where 
RTPProxy is behind NAT and the address inidcated is the private one. You 
should swap this IP with the public advertised one.


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/08/2016 09:51 PM, Robert Dyck wrote:

Thank you

Assuming rtpproxy was started with IPV4 as the first address and IPV6 as the
second, then in the NAT scenario, are the II flags mandatory in offer/answer?

Slightly off topic, what sort of scenario would require the address parameter
for offer/answer?

On November 8, 2016 09:57:30 AM Răzvan Crainea wrote:

Hi, Robert!

See my answers inline.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/08/2016 02:15 AM, Robert Dyck wrote:

I have some question regarding rtpproxy capabilities in relation to
IPV4-IPV6 interworking.

The articles I have read say that you need to assign an address from each
address family to rtpproxy. They go on to say that rtpproxy will then be
in
bridged mode. Others define bridge mode as assigning two interfaces to
rtpproxy.

As long as you have RTPProxy listening on two IPs, you have it set in
bridge mode. It doesn't matther whether one of them is IPv6, or both are.


If the IPV4 and IPV6 addresses are on the same interface, is the rtpproxy
indeed in bridged mode? Should one avoid the use of engage_rtpproxy?

Yes, as stated above, RTPProxy is in bridged mode and you should avoid
using engage_rtpproxy(). That's because the function can't know/decide
which interface is which and cannot map with the RTPProxy's one.


Assuming that IPV4- IPV6 interworking is actually possible using opensips
and rtpproxy, does that mean that an instance of rtpproxy is not
available to enable NAT traversal - would NAT traversal require using
another instance of rtpproxy using a single IPV4 address?

No, you don't need an extra instance - a single instance will do both
bridging and nat traversal.


Furthermore is the multihome parameter relevant to IPV4-IPV6 interworking
if opensips only listens on one interface?

The multihome parameter is only relevant for OpenSIPS, it doesn't
influence RTPProxy's behavior at all.

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Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

[Robert Dyck]

Thank you

Assuming rtpproxy was started with IPV4 as the first address and IPV6 as the 
second, then in the NAT scenario, are the II flags mandatory in offer/answer?

Slightly off topic, what sort of scenario would require the address parameter 
for offer/answer?

On November 8, 2016 09:57:30 AM Răzvan Crainea wrote:
> Hi, Robert!
> 
> See my answers inline.
> 
> Best regards,
> 
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> 
> On 11/08/2016 02:15 AM, Robert Dyck wrote:
> > I have some question regarding rtpproxy capabilities in relation to
> > IPV4-IPV6 interworking.
> > 
> > The articles I have read say that you need to assign an address from each
> > address family to rtpproxy. They go on to say that rtpproxy will then be
> > in
> > bridged mode. Others define bridge mode as assigning two interfaces to
> > rtpproxy.
> 
> As long as you have RTPProxy listening on two IPs, you have it set in
> bridge mode. It doesn't matther whether one of them is IPv6, or both are.
> 
> > If the IPV4 and IPV6 addresses are on the same interface, is the rtpproxy
> > indeed in bridged mode? Should one avoid the use of engage_rtpproxy?
> 
> Yes, as stated above, RTPProxy is in bridged mode and you should avoid
> using engage_rtpproxy(). That's because the function can't know/decide
> which interface is which and cannot map with the RTPProxy's one.
> 
> > Assuming that IPV4- IPV6 interworking is actually possible using opensips
> > and rtpproxy, does that mean that an instance of rtpproxy is not
> > available to enable NAT traversal - would NAT traversal require using
> > another instance of rtpproxy using a single IPV4 address?
> 
> No, you don't need an extra instance - a single instance will do both
> bridging and nat traversal.
> 
> > Furthermore is the multihome parameter relevant to IPV4-IPV6 interworking
> > if opensips only listens on one interface?
> 
> The multihome parameter is only relevant for OpenSIPS, it doesn't
> influence RTPProxy's behavior at all.
> 
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Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

[Răzvan Crainea]

Hi, Robert!

See my answers inline.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/08/2016 02:15 AM, Robert Dyck wrote:

I have some question regarding rtpproxy capabilities in relation to IPV4-IPV6
interworking.

The articles I have read say that you need to assign an address from each
address family to rtpproxy. They go on to say that rtpproxy will then be in
bridged mode. Others define bridge mode as assigning two interfaces to
rtpproxy.
As long as you have RTPProxy listening on two IPs, you have it set in 
bridge mode. It doesn't matther whether one of them is IPv6, or both are.

If the IPV4 and IPV6 addresses are on the same interface, is the rtpproxy
indeed in bridged mode? Should one avoid the use of engage_rtpproxy?
Yes, as stated above, RTPProxy is in bridged mode and you should avoid 
using engage_rtpproxy(). That's because the function can't know/decide 
which interface is which and cannot map with the RTPProxy's one.


Assuming that IPV4- IPV6 interworking is actually possible using opensips and
rtpproxy, does that mean that an instance of rtpproxy is not available to
enable NAT traversal - would NAT traversal require using another instance of
rtpproxy using a single IPV4 address?
No, you don't need an extra instance - a single instance will do both 
bridging and nat traversal.


Furthermore is the multihome parameter relevant to IPV4-IPV6 interworking if
opensips only listens on one interface?
The multihome parameter is only relevant for OpenSIPS, it doesn't 
influence RTPProxy's behavior at all.


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[OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

[Robert Dyck]

I have some question regarding rtpproxy capabilities in relation to IPV4-IPV6 
interworking.

The articles I have read say that you need to assign an address from each 
address family to rtpproxy. They go on to say that rtpproxy will then be in 
bridged mode. Others define bridge mode as assigning two interfaces to 
rtpproxy.
If the IPV4 and IPV6 addresses are on the same interface, is the rtpproxy 
indeed in bridged mode? Should one avoid the use of engage_rtpproxy?

Assuming that IPV4- IPV6 interworking is actually possible using opensips and 
rtpproxy, does that mean that an instance of rtpproxy is not available to 
enable NAT traversal - would NAT traversal require using another instance of 
rtpproxy using a single IPV4 address?

Furthermore is the multihome parameter relevant to IPV4-IPV6 interworking if 
opensips only listens on one interface?

Thank you all

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Re: [OpenSIPS-Users] rtpproxy TCP connection

[Nabeel]

Could this also explain why other TURN servers don't work so well over TCP
with OpenSIPS?


On 18 February 2016 at 12:09, Răzvan Crainea  wrote:

> Hi Gomtesh!
>
> Currently only UDP and UNIX datagrams sockets are supported.
>
> Best regards,
> Răzvan
>
>
> On 02/18/2016 01:15 PM, Gomtesh Jain wrote:
>
> Is it possible to make tcp connection to rtpproxy from opensips ? I am
> trying
>  modparam("rtpproxy", "rtpproxy_sock", "tcp:127.0.0.1:2")
>
> But it is not working .
> any suggestions ?
>
> Thanks,
> Gomtesh
>
>
>
> ___
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>
> --
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> OpenSIPS Core Developerhttp://www.opensips-solutions.com
>
>
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Re: [OpenSIPS-Users] rtpproxy TCP connection

[Răzvan Crainea]

Hi Gomtesh!

Currently only UDP and UNIX datagrams sockets are supported.

Best regards,
Răzvan

On 02/18/2016 01:15 PM, Gomtesh Jain wrote:
Is it possible to make tcp connection to rtpproxy from opensips ? I am 
trying
modparam("rtpproxy", "rtpproxy_sock", "tcp:127.0.0.1:2 
")


But it is not working .
any suggestions ?

Thanks,
Gomtesh



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http://www.opensips-solutions.com

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[OpenSIPS-Users] rtpproxy TCP connection

[Gomtesh Jain]

Is it possible to make tcp connection to rtpproxy from opensips ? I am
trying
 modparam("rtpproxy", "rtpproxy_sock", "tcp:127.0.0.1:2")

But it is not working .
any suggestions ?

Thanks,
Gomtesh
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Re: [OpenSIPS-Users] rtpproxy and parallel forking

[Bogdan-Andrei Iancu]

Hi Pete,

I assume you do rtpproxy_answer() for the 200 OK on B leg, right ?

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 23.09.2015 11:44, Pete Kelly wrote:
I am using rtpproxy with parallel fork and noticed some interesting 
behaviour (by rtpproxy).


If the INVITE is forked to 2 destinations (A and B), one of them (A) 
may send a 183 with media, meaning there is media being sent to the 
rtpproxy.


However if it is B that answers, rtpproxy will still only be set up to 
send and receive media to A, and will continue to do so which means 
there is no media on the call.


Reading the rtpproxy docs I think it is because of this:

"After the session has been created, the proxy listens on the port it 
has allocated for that session and waits for receiving at least one 
UDP packet from each of two parties participating in the call. Once 
such packet is received, the proxy fills one of two ip:port structures 
associated with each call with source ip:port of that packet"


Is there a known way round this issue, other than stopping A from 
sending media to rtpproxy or using late offer INVITEs?



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Re: [OpenSIPS-Users] rtpproxy and parallel forking

[Eric Tamme]

Hi Pete,

To the best of my knowledge no rtp proxy: mediarelay, rtpengine, 
rtpproxy deals with forking and early media "well".  I believe this is 
more a failing of the 183 draft than anything else.  For example If I 
parallel fork a call to A and B, A sends 183 with an IVR but then B 
sends a 200 it is not clear what should be done - send a CANCEL to A and 
terminate any IVR?


RTPEngine does have ... sort of a work around in that it will allow you 
to specify whether or not to automatically "train" to a rtp source - 
this allows you to set up a call with early media to A, but then if B 
starts sending RTP to the same allotted ports RTPEngine will simply 
switch to those ports.  This has several security implications - 
Freeswitch has a similar feature which allows the rtp source to change 
within a given allotted "buffer".


To answer your question directly - no, I do not know of a way to do 
parallel forking with rtpproxy where one leg may send early media. We 
have experienced this as well when our customers put multiple pstn phone 
numbers in a ring group and have advised them that it will not work 
should one of those numbers provide early media.


Hope all is well,
-Eric



On 09/23/2015 02:44 AM, Pete Kelly wrote:
I am using rtpproxy with parallel fork and noticed some interesting 
behaviour (by rtpproxy).


If the INVITE is forked to 2 destinations (A and B), one of them (A) 
may send a 183 with media, meaning there is media being sent to the 
rtpproxy.


However if it is B that answers, rtpproxy will still only be set up to 
send and receive media to A, and will continue to do so which means 
there is no media on the call.


Reading the rtpproxy docs I think it is because of this:

"After the session has been created, the proxy listens on the port it 
has allocated for that session and waits for receiving at least one 
UDP packet from each of two parties participating in the call. Once 
such packet is received, the proxy fills one of two ip:port structures 
associated with each call with source ip:port of that packet"


Is there a known way round this issue, other than stopping A from 
sending media to rtpproxy or using late offer INVITEs?



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[OpenSIPS-Users] rtpproxy and parallel forking

[Pete Kelly]

I am using rtpproxy with parallel fork and noticed some interesting
behaviour (by rtpproxy).

If the INVITE is forked to 2 destinations (A and B), one of them (A) may
send a 183 with media, meaning there is media being sent to the rtpproxy.

However if it is B that answers, rtpproxy will still only be set up to send
and receive media to A, and will continue to do so which means there is no
media on the call.

Reading the rtpproxy docs I think it is because of this:

"After the session has been created, the proxy listens on the port it has
allocated for that session and waits for receiving at least one UDP packet
from each of two parties participating in the call. Once such packet is
received, the proxy fills one of two ip:port structures associated with
each call with source ip:port of that packet"

Is there a known way round this issue, other than stopping A from sending
media to rtpproxy or using late offer INVITEs?
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Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

[Nabeel]

Thanks, but I'm still looking for a more direct comparison of rtpproxy vs.
TURN/ICE only based on their effectiveness, nothing else.

I know both work but I would like to know of any evidence that TURN with
two public IPs is more effective than rtpproxy alone.
On 29 Aug 2015 18:36, "Giovanni Maruzzelli"  wrote:

> Both will work.
>
> You can check other aspects inherently to your project and implementation:
> performances, integration, etc
>
> Rttproxy, media engine and the like can give you more services related to
> the fact they are controlled by the proxy.
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Aug 29, 2015 7:04 PM, "Nabeel"  wrote:
>
>> Sorry previous message I sent was meant to be a quote.
>>
>> All my clients will use the same UAC which supports ICE/TURN, so that is
>> not an issue.
>>
>> I just want to know which is more effective solely on the basis of NAT
>> traversal ability.
>> On 29 Aug 2015 18:01, "Nabeel"  wrote:
>>
>>> That said, only clients that supports turn will use it, check your
>>> clients features.
>>>
>>> Rtpproxy, mediaengine, and the like do not rely on clients support, they
>>> are.enforced by sip proxy manipulation of sdp.
>>> On 29 Aug 2015 17:02, "Giovanni Maruzzelli"  wrote:
>>>
 Stun/turn are the only methods used by webrtc peers, and because are
 used through ICE they're very effective.

 You can check coturn for an advanced implementation.

 That said, only clients that supports turn will use it, check your
 clients features.

 Rtpproxy, mediaengine, and the like do not rely on clients support,
 they are.enforced by sip proxy manipulation of sdp.

 So, actually they (turn and rtpproxy) are not alternative to each
 other, but complementary.
 Eg: your service can offer both technologies at the same time, clients
 choose what to do.

 -giovanni

 sent from my mobile,
 Giovanni Maruzzelli
 cell: +39 347 266 56 18
 On Aug 29, 2015 5:48 PM, "Nabeel"  wrote:

> Hi,
>
> I would like to know which is more effective for NAT traversal,
> rtpproxy or STUN/TURN/ICE implementation.
>
> I heard that TURN server with one public IP can function equivalent to
> rtpproxy, and TURN server with two public IPs is more effective than
> rtpproxy.
>
> Is that true?
>
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 ___
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 Users@lists.opensips.org
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Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

[Giovanni Maruzzelli]

Both will work.

You can check other aspects inherently to your project and implementation:
performances, integration, etc

Rttproxy, media engine and the like can give you more services related to
the fact they are controlled by the proxy.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Aug 29, 2015 7:04 PM, "Nabeel"  wrote:

> Sorry previous message I sent was meant to be a quote.
>
> All my clients will use the same UAC which supports ICE/TURN, so that is
> not an issue.
>
> I just want to know which is more effective solely on the basis of NAT
> traversal ability.
> On 29 Aug 2015 18:01, "Nabeel"  wrote:
>
>> That said, only clients that supports turn will use it, check your
>> clients features.
>>
>> Rtpproxy, mediaengine, and the like do not rely on clients support, they
>> are.enforced by sip proxy manipulation of sdp.
>> On 29 Aug 2015 17:02, "Giovanni Maruzzelli"  wrote:
>>
>>> Stun/turn are the only methods used by webrtc peers, and because are
>>> used through ICE they're very effective.
>>>
>>> You can check coturn for an advanced implementation.
>>>
>>> That said, only clients that supports turn will use it, check your
>>> clients features.
>>>
>>> Rtpproxy, mediaengine, and the like do not rely on clients support, they
>>> are.enforced by sip proxy manipulation of sdp.
>>>
>>> So, actually they (turn and rtpproxy) are not alternative to each other,
>>> but complementary.
>>> Eg: your service can offer both technologies at the same time, clients
>>> choose what to do.
>>>
>>> -giovanni
>>>
>>> sent from my mobile,
>>> Giovanni Maruzzelli
>>> cell: +39 347 266 56 18
>>> On Aug 29, 2015 5:48 PM, "Nabeel"  wrote:
>>>
 Hi,

 I would like to know which is more effective for NAT traversal,
 rtpproxy or STUN/TURN/ICE implementation.

 I heard that TURN server with one public IP can function equivalent to
 rtpproxy, and TURN server with two public IPs is more effective than
 rtpproxy.

 Is that true?

 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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>>>
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Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

[Nabeel]

Sorry previous message I sent was meant to be a quote.

All my clients will use the same UAC which supports ICE/TURN, so that is
not an issue.

I just want to know which is more effective solely on the basis of NAT
traversal ability.
On 29 Aug 2015 18:01, "Nabeel"  wrote:

> That said, only clients that supports turn will use it, check your clients
> features.
>
> Rtpproxy, mediaengine, and the like do not rely on clients support, they
> are.enforced by sip proxy manipulation of sdp.
> On 29 Aug 2015 17:02, "Giovanni Maruzzelli"  wrote:
>
>> Stun/turn are the only methods used by webrtc peers, and because are used
>> through ICE they're very effective.
>>
>> You can check coturn for an advanced implementation.
>>
>> That said, only clients that supports turn will use it, check your
>> clients features.
>>
>> Rtpproxy, mediaengine, and the like do not rely on clients support, they
>> are.enforced by sip proxy manipulation of sdp.
>>
>> So, actually they (turn and rtpproxy) are not alternative to each other,
>> but complementary.
>> Eg: your service can offer both technologies at the same time, clients
>> choose what to do.
>>
>> -giovanni
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On Aug 29, 2015 5:48 PM, "Nabeel"  wrote:
>>
>>> Hi,
>>>
>>> I would like to know which is more effective for NAT traversal, rtpproxy
>>> or STUN/TURN/ICE implementation.
>>>
>>> I heard that TURN server with one public IP can function equivalent to
>>> rtpproxy, and TURN server with two public IPs is more effective than
>>> rtpproxy.
>>>
>>> Is that true?
>>>
>>> ___
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>>>
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Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

[Nabeel]

That said, only clients that supports turn will use it, check your clients
features.

Rtpproxy, mediaengine, and the like do not rely on clients support, they
are.enforced by sip proxy manipulation of sdp.
On 29 Aug 2015 17:02, "Giovanni Maruzzelli"  wrote:

> Stun/turn are the only methods used by webrtc peers, and because are used
> through ICE they're very effective.
>
> You can check coturn for an advanced implementation.
>
> That said, only clients that supports turn will use it, check your clients
> features.
>
> Rtpproxy, mediaengine, and the like do not rely on clients support, they
> are.enforced by sip proxy manipulation of sdp.
>
> So, actually they (turn and rtpproxy) are not alternative to each other,
> but complementary.
> Eg: your service can offer both technologies at the same time, clients
> choose what to do.
>
> -giovanni
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Aug 29, 2015 5:48 PM, "Nabeel"  wrote:
>
>> Hi,
>>
>> I would like to know which is more effective for NAT traversal, rtpproxy
>> or STUN/TURN/ICE implementation.
>>
>> I heard that TURN server with one public IP can function equivalent to
>> rtpproxy, and TURN server with two public IPs is more effective than
>> rtpproxy.
>>
>> Is that true?
>>
>> ___
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>>
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Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

[Stefano Pisani]

Stun/Turn/Ice are usefull where Client is behind a NAT and OpenSIPS has 
public IP.
You could use nathelper modules instead of Stun, to set the right IPs in 
the messages from client.
If OpenSIPS is behind a NAT too (it has private IP) you must use 
RTPProxy too with a proper configuration.



Il 29/08/2015 17:47, Nabeel ha scritto:


Hi,

I would like to know which is more effective for NAT traversal, 
rtpproxy or STUN/TURN/ICE implementation.


I heard that TURN server with one public IP can function equivalent to 
rtpproxy, and TURN server with two public IPs is more effective than 
rtpproxy.


Is that true?



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Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

[Giovanni Maruzzelli]

Stun/turn are the only methods used by webrtc peers, and because are used
through ICE they're very effective.

You can check coturn for an advanced implementation.

That said, only clients that supports turn will use it, check your clients
features.

Rtpproxy, mediaengine, and the like do not rely on clients support, they
are.enforced by sip proxy manipulation of sdp.

So, actually they (turn and rtpproxy) are not alternative to each other,
but complementary.
Eg: your service can offer both technologies at the same time, clients
choose what to do.

-giovanni

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Aug 29, 2015 5:48 PM, "Nabeel"  wrote:

> Hi,
>
> I would like to know which is more effective for NAT traversal, rtpproxy
> or STUN/TURN/ICE implementation.
>
> I heard that TURN server with one public IP can function equivalent to
> rtpproxy, and TURN server with two public IPs is more effective than
> rtpproxy.
>
> Is that true?
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

[Nabeel]

Hi,

I would like to know which is more effective for NAT traversal, rtpproxy or
STUN/TURN/ICE implementation.

I heard that TURN server with one public IP can function equivalent to
rtpproxy, and TURN server with two public IPs is more effective than
rtpproxy.

Is that true?
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Re: [OpenSIPS-Users] [RTPproxy] Re: Announcing rtpproxy v2.0.0

[Maxim Sobolev]

Pull requests are welcome, guys. We are hitting 30k rtp sessions on one of
our largest clusters. Lot of changes are coming into git repo soon to make
rtp_cluster component run smoothly under such conditions.
On Mar 16, 2015 8:56 PM, "John Mathew"  wrote:

> Yes
>
> On Tuesday, 17 March 2015, Zheng Frank  wrote:
>
>> Do you mean ROHC ?
>>
>> 2015-03-14 12:39 GMT+08:00 Maxim Sobolev :
>>
>>> Do you have any particular RFC in mind?
>>> On Mar 12, 2015 10:28 AM, "John Mathew" 
>>> wrote:
>>>
 Hi,

 Maxim,
 Is there any plans for rtp header compression in future. I can't see
 anything in the change log for 2.0.0


 On Tuesday, 10 March 2015, Maxim Sobolev  wrote:

> Hi All,
>
> I'm happy to announce that we have released rtpproxy v2.0.0.
>
> You can review the release notes here:
> https://github.com/sippy/rtpproxy/releases/tag/v2.0.0
>
> -sobomax
>
>

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Re: [OpenSIPS-Users] [RTPproxy] Announcing rtpproxy v2.0.0

[Mete Gonenc]

+1 That would be a great feature.

On Thursday, March 12, 2015, John Mathew  wrote:

> Hi,
>
> Maxim,
> Is there any plans for rtp header compression in future. I can't see
> anything in the change log for 2.0.0
>
> On Tuesday, 10 March 2015, Maxim Sobolev  > wrote:
>
>> Hi All,
>>
>> I'm happy to announce that we have released rtpproxy v2.0.0.
>>
>> You can review the release notes here:
>> https://github.com/sippy/rtpproxy/releases/tag/v2.0.0
>>
>> -sobomax
>>
>>
>
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Re: [OpenSIPS-Users] [RTPproxy] Re: Announcing rtpproxy v2.0.0

[Maxim Sobolev]

Do you have any particular RFC in mind?
On Mar 12, 2015 10:28 AM, "John Mathew"  wrote:

> Hi,
>
> Maxim,
> Is there any plans for rtp header compression in future. I can't see
> anything in the change log for 2.0.0
>
> On Tuesday, 10 March 2015, Maxim Sobolev  wrote:
>
>> Hi All,
>>
>> I'm happy to announce that we have released rtpproxy v2.0.0.
>>
>> You can review the release notes here:
>> https://github.com/sippy/rtpproxy/releases/tag/v2.0.0
>>
>> -sobomax
>>
>>
>
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Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK

[Răzvan Crainea]

Hi Marco!

As Patrick suggested, adding the a:sendonly line in RTP should instruct 
the caller not to send any RTP. However, if I remember correctly, I've 
seen legitimate clients that still send RTP.
On a different note, they are sending RTP to a media gateway, right? And 
most likely the B part will ignore all the RTP.


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 01/22/2015 02:50 PM, Patrick Wakano wrote:

Ok Marco,
Your concern is with hackers and not misuse! Really valid nowadays!

Patrick

On Thu, Jan 22, 2015 at 8:32 AM, Marco Hierl 
mailto:marco.hi...@mrnetgroup.com>> wrote:


Hi Patrik,

thanks for this idea!

I did not say clear enough: I’m afraid that anybody can cheat us.
My intention is to assure that our interconnection partners (or
their customers) do not have the possibility to make a
conversation without being charged.

Sending the indication “a:sendonly” only means, that the client is
told not to send RTP, but IF it send RTP anyway then the RTPproxy
leads in on to the callee. So, it is not in my hands then!

Best regards from Hamburg

  Marco

*Von:*users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>
[mailto:users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>] *Im Auftrag von
*Patrick Wakano
*Gesendet:* Donnerstag, 22. Januar 2015 11:16
*An:* OpenSIPS users mailling list

*Betreff:* Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to
callee before 200OK

Have you tried to insert a "a:sendonly" line in your SDP body when
sending it to the caller?
If the client receives such line it should not send media...

Then in the 200Ok you can put an "a:sendrecv" line to establish
full media path!

It's just an idea, I'm not sure if it will really work...

Patrick

On Thu, Jan 22, 2015 at 6:51 AM, Marco Hierl
mailto:marco.hi...@mrnetgroup.com>>
wrote:

Hi Răzvan,

Ok, thanks for your answer!

Unfortunately we are offering „early media“ to our customers (call
center, radio station, and other companies) and lots of them like
to play a free-of-charge announcement in the beginning. But if we
started to get cheated, maybe we need to go for this workaround.

But apart from that: Mostly the SDP is NOT repeated in the 200OK.
Can I call rtpproxy_answer() when receiving the 200OK anyway?

Thanks and best regards

  Marco

*Von:*users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>
[mailto:users-boun...@lists.opensips.org
<mailto:users-boun...@lists.opensips.org>] *Im Auftrag von *Razvan
Crainea
*Gesendet:* Donnerstag, 22. Januar 2015 09:36
*An:* users@lists.opensips.org <mailto:users@lists.opensips.org>
*Betreff:* Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to
callee before 200OK

Hi, Marco!

From RTPProxy point of view, you can't differentiate between SIP
replies, because for all of them you call the same function -
rtpproxy_answer().
Now, if the client decides to send RTP for 183 (and indeed, I've
seen this several times), there's not that much that you can do.
Although it's kind of a hack, all I can think of is to not call
rtpproxy_answer() for 180/183 and strip the body to prevent the
client from sending RTP directly to the callee.
I hope this works for you.

Best regards,

Răzvan Crainea

OpenSIPS Solutions

www.opensips-solutions.com  <http://www.opensips-solutions.com>

On 01/21/2015 04:07 PM, Marco Hierl wrote:

Dear all,

first of all I need to apologize that I was not able to find
information about this issue although I’m sure that I’m not
the first one complaining!

The caller is sending an INVITE via OpenSIPS and
rtpproxy_offer() is executed, callee answers with REPLY 180 or
REPLY 183 (with SDP) and rtpproxy_answer() is made. In this
status it should be ok that the rtp stream from callee to
caller is transferred via the rtpproxy (e.g. for
announcements), but I can see that rtp stream from caller to
callee is transferred too!!! This means that there can be a
conversation without receiving the 200OK and what is the real
problem: that means (at least for me) they can talk to each
other without any charging !! A timer will stop the conversion
after the a while, but this can take time.

How can I overcome this problem? How can prevent RTP to be
send to the callee before REPLY 200 is received?

I can’t find any help in the RTPproxy protocol
http://www.b2bua.org/wiki/RTPproxy/Protocol, nor in the
rtpproxy module description in OpenSIPS.

Thanks for your ideas, and best reg