Re: [SR-Users] rtpproxy doesn't record after start_recording

[Marko Tirs]

Hi Daniel,


yes, I hear the audio going both ways ok.

In the meantime I succeeded to record something by rtpproxy
but in best case only one direction, I can hear just the callee,
but not the caller.

In the meantime I have used start_recording in route[NATMANAGE]
but before rtpproxy_manage("co"):
--

if (is_method("INVITE") and (status=="200")) {
   start_recording();
} 

rtpproxy_manage("co");
-- 


After reading of your message I tested also with

start_recording after rtpproxy_manage("co"):
-- 
rtpproxy_manage("co");

if (is_method("INVITE") and (status=="200")) {
   start_recording();
}
-- 
But the result was the same:

I have 2 SIP users:
31 (X-Lite on Windows, static address 192.168.0.11)
35 (Android Zoiper, IP address 192.168.0.29 (by DHCP))
Kamailio has static address 192.168.0.13.

When 35 calls 31 then I hear 1 direction in the recorded rtp file,
the voice of callee.
When 31 calls 35 then I don't hear anything in the recorded file.

I don't understand why this difference when user 31 or 35 initiates
a call!?

But I hear audio conversation in both directions, just from the
recorded file in one direction, or even nothing.

I get 2 files per one call: .rtp and .rtcp
How many files should rtpproxy produce for one call?
Should also a file or two for 2nd channel be generated?


How can I record both audio ways?

Thank you
Marko




The whole function route[NATMANAGE]:



# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;


if (is_method("INVITE") and (status=="200")) {
   start_recording();
}

rtpproxy_manage("co");

if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
set_contact_alias();
}
}
#!endif
return;
}








- Original Message -----
From: Daniel-Constantin Mierla 
To: Marko Tirs ; Kamailio (SER) - Users Mailing List 

Sent: Tuesday, March 14, 2017 5:23 PM
Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording

Hello,

I think you can call the start recording function just after
rtpproxy_manage().

If you don't use the start recording function, is the audio going both
ways without problems?

Cheers,
Daniel

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Re: [SR-Users] rtpproxy doesn't record after start_recording

[Marko Tirs]

Hi Daniel,


yes, I hear the audio going both ways ok.

In the meantime I succeeded to record something by rtpproxy
but in best case only one direction, I can hear just the callee,
but not the caller.

In the meantime I have used start_recording in route[NATMANAGE]
but before rtpproxy_manage("co"):

--

if (is_method("INVITE") and (status=="200")) {
   start_recording();
} 

rtpproxy_manage("co");
-- 


After reading of your message I tested also with

start_recording after rtpproxy_manage("co"):
-- 

rtpproxy_manage("co");

if (is_method("INVITE") and (status=="200")) {
   start_recording();
}

-- 
But the result was the same:

I have 2 SIP users:
31 (X-Lite on Windows, static address 192.168.0.11)
35 (Android Zoiper, IP address 192.168.0.29 (by DHCP))
Kamailio has static address 192.168.0.13.

When 35 calls 31 then I hear 1 direction in the recorded rtp file,
the voice of callee.
When 31 calls 35 then I don't hear anything in the recorded file.

I don't understand why this difference when user 31 or 35 initiates
a call!?

But I hear audio conversation in both directions, just from the
recorded file in one direction, or even nothing.

I get 2 files per one call: .rtp and .rtcp
How many files should rtpproxy produce for one call?
Should also a file or two for 2nd channel be generated?


How can I record both audio ways?

Thank you
Marko




The whole function route[NATMANAGE]:



# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;


if (is_method("INVITE") and (status=="200")) {
   start_recording();
}

rtpproxy_manage("co");

if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
set_contact_alias();
}
}
#!endif
return;
}








- Original Message -----
From: Daniel-Constantin Mierla 
To: Marko Tirs ; Kamailio (SER) - Users Mailing List 

Sent: Tuesday, March 14, 2017 5:23 PM
Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording

Hello,

I think you can call the start recording function just after
rtpproxy_manage().

If you don't use the start recording function, is the audio going both
ways without problems?

Cheers,
Daniel

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Re: [SR-Users] rtpproxy doesn't record after start_recording

[Daniel-Constantin Mierla]

Hello,

I think you can call the start recording function just after
rtpproxy_manage().

If you don't use the start recording function, is the audio going both
ways without problems?

Cheers,
Daniel


On 12/03/2017 15:04, Marko Tirs wrote:
> Hello experts,
>
> I'm newbie in Kamailio and have a problem with my 1st installation.
> I just want to control the calls between SIP clients registered by Kamailio
> and record the conversations.
>
> I installed rtpproxy and started with
> rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.0.13 -r /home/user1/rtpproxy
> -d DBUG:LOG_LOCAL0
>
> After start_recording() in the function onreply_route[MANAGE_REPLY]
> rtpproxy creates the files .rtp and .rtcp but they
> remain at zero length.
>
> 2nd problem is that the media stream is broken in one direction
> after start_recording() has been executed, the callee doesn't hear
> the caller.
>
> How is the right Kamailio configuration for recording conversations?
>
> Thank you
> Regards
> Marko
>
>
> Modified kamailio-basic.cfg partly:
>
> #!KAMAILIO
>
> #!define WITH_NAT
>
> #!ifdef WITH_NAT
> loadmodule "nathelper.so"
> loadmodule "rtpproxy.so"
> #!endif
>
> #!ifdef WITH_NAT
> # - rtpproxy params -
> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
>
> # - nathelper params -
> modparam("nathelper", "natping_interval", 30)
> modparam("nathelper", "ping_nated_only", 1)
> modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
> modparam("nathelper", "sipping_from", "sip:pin...@kamailio.org")
>
> # params needed for NAT traversal in other modules
> modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
> modparam("usrloc", "nat_bflag", FLB_NATB)
> #!endif
>
> ### Routing Logic 
>
> route[RELAY] {
> # enable additional event routes for forwarded requests
> # - serial forking, RTP relaying handling, a.s.o.
> if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
> if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
> }
> if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
> if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
> }
> if (is_method("INVITE")) {
> if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
> }
>
> if (!t_relay()) {
> sl_reply_error();
> }
> exit;
> }
>
> # RTPProxy control
> route[NATMANAGE] {
> #!ifdef WITH_NAT
> if (is_request()) {
> if(has_totag()) {
> if(check_route_param("nat=yes")) {
> setbflag(FLB_NATB);
> }
> }
> }
> if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
> return;
>
> rtpproxy_manage("co");
>
> if (is_request()) {
> if (!has_totag()) {
> if(t_is_branch_route()) {
> add_rr_param(";nat=yes");
> }
> }
> }
> if (is_reply()) {
> if(isbflagset(FLB_NATB)) {
> set_contact_alias();
> }
> }
> #!endif
> return;
> }
>
> # Manage incoming replies
> onreply_route[MANAGE_REPLY] {
> xdbg("incoming reply\n");
> if (is_method("INVITE") and (status=="200")) {
> xlog("START_RECORDING #1\n");
> start_recording();
> xlog("START_RECORDING #2\n");
> }
> else if(status=~"[12][0-9][0-9]")
> route(NATMANAGE);
> }
>
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-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - 
www.asipto.com
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com


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Re: [SR-Users] RTPProxy with VPN

[Daniel-Constantin Mierla]

Hello,

you need to run rtpproxy in bridge mode. See the alg example from the
rtpproxy module -- online at:

  -
https://github.com/kamailio/kamailio/blob/master/src/modules/rtpproxy/examples/alg.cfg

Cheers,
Daniel


On 05/01/2017 01:09, Rodrigo Moreira wrote:
> Hello,
>
> I am having problems with RTPPROXY. I need it to work as a bridge. On
> one side will have a VPN tunnel called a tun0, on the other an eth0
> interface. Is it possible for RTPPROXY to run correctly in this scenario?
>
> How can I test if RTPPROXY is running correctly? Is there any command?
> What would the configuration look like inside /etc/default/rtpproxy
>
> Please help me.
>
> Best regards
> -- 
> Rodrigo M.
> (37) 9132-4539
> (34) 9889-3069
> rodrigo.moreira2007
>
>

-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Re: [SR-Users] rtpproxy/rtpengine. Help to understand.

[Igor Olhovskiy]

Yep, logs helps. Actually, was mix of errors in PBX and Kamailio config.
So yea, rtpproxy is working as expected, can’t say same for my config files :)

Regards, Igor

On 22 дек. 2016 г., 19:24 +0200, Alex Balashov , 
wrote:
> That just sounds like the rtpproxy is not being engaged, i.e. that the 
> rtpproxy_manage() call is failing. When that happens, the SDP from .2 will be 
> passed through unaltered.
>
> The Kamailio log should give you some idea of why the rtpproxy invocation has 
> failed.
>
> RTPProxy is certainly not limited to RFC 1918 addresses or any particular 
> network characteristics.
>
> -- Alex
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Re: [SR-Users] rtpproxy/rtpengine. Help to understand.

[Alex Balashov]

Oh, I see. Yes, that could be. 

-- Alex

> On Dec 22, 2016, at 12:49 PM, Daniel Tryba  wrote:
> 
>> On Thu, Dec 22, 2016 at 12:23:52PM -0500, Alex Balashov wrote:
>> That just sounds like the rtpproxy is not being engaged, i.e. that the
>> rtpproxy_manage() call is failing. When that happens, the SDP from .2
>> will be passed through unaltered.
>> 
>> The Kamailio log should give you some idea of why the rtpproxy
>> invocation has failed.
> 
> I'd guess rtpproxy_manage() isn't failing, it just isn't called since
> dialplan logic is only instructed to do this for natted stuff. Since the
> source ip and SDP both contain the same ip the assumption is that
> rtpproxy is not needed.
> 
> 
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Re: [SR-Users] rtpproxy/rtpengine. Help to understand.

[Daniel Tryba]

On Thu, Dec 22, 2016 at 12:23:52PM -0500, Alex Balashov wrote:
> That just sounds like the rtpproxy is not being engaged, i.e. that the
> rtpproxy_manage() call is failing. When that happens, the SDP from .2
> will be passed through unaltered.
> 
> The Kamailio log should give you some idea of why the rtpproxy
> invocation has failed.

I'd guess rtpproxy_manage() isn't failing, it just isn't called since
dialplan logic is only instructed to do this for natted stuff. Since the
source ip and SDP both contain the same ip the assumption is that
rtpproxy is not needed.


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Re: [SR-Users] rtpproxy/rtpengine. Help to understand.

[Alex Balashov]

That just sounds like the rtpproxy is not being engaged, i.e. that the 
rtpproxy_manage() call is failing. When that happens, the SDP from .2 will be 
passed through unaltered.

The Kamailio log should give you some idea of why the rtpproxy invocation has 
failed.

RTPProxy is certainly not limited to RFC 1918 addresses or any particular 
network characteristics.

-- Alex
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Re: [SR-Users] rtpproxy/rtpengine. Help to understand.

[Igor Olhovskiy]

Sorry,  I got media address 111.222.3.2 in SDP, but not 111.222.3.3

Regards, Igor

On 22 дек. 2016 г., 19:19 +0200, Igor Olhovskiy , 
wrote:
> Hi!
> Issue I can’t figure out. Or all working ok and that’s just me who not 
> understands.
>
> I have situation
> softPhone (111.222.3.2) -> Kamailio w. rtpproxy (111.222.3.3) -> PBX 
> (111.222.3.4)
> All addresses are on same public network.
> rtpproxy is running with -l 111.222.3.3 -A 111.222.3.3
> I want all media goes through rtpproxy. So, in INVITE and ACK I call 
> rtpproxy_manage().
> But according to sip packets on PBX (10.0.0.4) I got media address 10.0.0.2 
> in SDP, but not 10.0.0.3
>
> If I use scheme, where client in some other NATted network (like 
> 192.168.1.100), all is replaced correctly (means rtpproxy_manage() is working)
>
> So, I can’t understand, rtpproxy_manage() changes only rfc1918 addresses or 
> I’m missing something?
> If any - can provide pcap’s with both cases.
>
> Regards, Igor
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Re: [SR-Users] Rtpproxy settings without NAT in Kamailio

[Vladyslav Zakhozhai]

Hi Shantanu,

You need to modify kamailio config to achive your goal. Look into NAT
routing (i.e. NATDETECT and NATMANAGE).

2016-12-03 8:23 GMT+02:00 shantanu saha :

> Hello,
> I am trying to configure rtpproxy with Kamailio. kamailio.cfg is already
> configured (default) for rtpproxy. I just add the following line at the top
> of the cfg file.
>
> *#!define WITH_NAT*
>
> After that rtpproxy is working in local network. Both client and sip
> server are running in local network.
> But when I run this in public network, rtpproxy didn't trigger.
>
> Rtppoxy and Kamailio are running in same public IP, Sip client also
> running in another *public IP. *So there is *no NAT* *.*
>
> Both local and public version was Centos 6
> Kamailio version : 4.4
> rtpproxy version: 2.1
>
> Note: udp connection between rtpproxy and Kamailio was established (both
> in public and private network)
>
>
> Thanks
> shantanu saha
>
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>


-- 
С уважением,
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Re: [SR-Users] RTPProxy benchmark

[Gholamreza Sabery]

Yes. I finish the session at the end of each call. Also I am using
rtpproxy_manage("r"). My configuration regarding RTPProxy is default config
which comes with Kamailio sample config file.

On Tue, Nov 15, 2016 at 11:44 AM, Dragos Oancea 
wrote:

> Hi
>
> Increase fs.file-max in your /etc/sysctl.conf .
> eg: fs.file-max = 5
> And then do sysctl -p
> Decrease SILENT_TIMEOUT in your rtpengine.conf (eg:SILENT_TIMEOUT=120) -
> it's default 1 hour and if some calls don't have media then rtpengine
> will just keep the UDP ports in use until this timeout expires.
>
> Regards,
> Dragos
>
>
> On 15/11/2016 09:01, Daniel-Constantin Mierla wrote:
> > Are you ending the rtp proxy sessions when the call are ended? What
> > rtpproxy functions are you using in the configuration file?
> >
> > Cheers,
> > Daniel
> >
> >
> > On 14/11/16 18:18, Gholamreza Sabery wrote:
> >> No not the first time. But over time. I rebooted my system and error
> >> is gone! It seems that it happens over time.
> >>
> >> On Mon, Nov 14, 2016 at 6:12 PM, Daniel-Constantin Mierla
> >> mailto:mico...@gmail.com>> wrote:
> >>
> >> Are you getting the error first time when you reach first 1900
> >> sessions? Or after a while, after some previous sessions are ended?
> >>
> >> Cheers,
> >> Daniel
> >>
> >>
> >> On 14/11/16 11:19, Gholamreza Sabery wrote:
> >>> I already set these parameters:
> >>>
> >>> rtpproxy -m 5000 -M 65000
> >>>
> >>> As well limits for number of open files are set to 100
> >>> (ulimit -n). When I increased log level of RTPProxy I saw:
> >>> ERR:create_twinlistener:GENERAL: can't create IPv4 socket: Too
> >>> many open files in system
> >>>
> >>>
> >>> On Mon, Nov 14, 2016 at 1:46 PM, Daniel-Constantin Mierla
> >>> mailto:mico...@gmail.com>> wrote:
> >>>
> >>> Hello,
> >>>
> >>> first thing to look at is the port range. There are some
> >>> parameter that you can provide to rtpproxy in command line in
> >>> order to increase the range of port it can use -- see
> >>> 'rtpproxy -h' or 'man rtpproxy'.
> >>>
> >>> Cheers,
> >>> Daniel
> >>>
> >>>
> >>> On 14/11/16 11:14, Gholamreza Sabery wrote:
>  Dear Daniel:
> 
>  I used a single RTPProxy instance. RTPProxy version =
>  20040107. And yes there was traffic for all calls but
>  traffic is one-way. One leg sends the call and the other
>  just receives it.
> 
>  On Mon, Nov 14, 2016 at 1:40 PM, Daniel-Constantin Mierla
>  mailto:mico...@gmail.com>> wrote:
> 
>  Hello,
> 
>  have you used a single rtpproxy instance? Was there RTP
>  traffic for all 1900 calls? Is this with rtpproxy 1.2 or
>  2.0?
> 
>  Cheers,
>  Daniel
> 
> 
>  On 14/11/16 10:44, Gholamreza Sabery wrote:
> > I managed to create about 1900 concurrent calls using a
> > single Kamailio and RTPProxy server. But after this
> > number RTPProxy returns 0 and the following error is
> > shown in the Kamailio log files:
> >
> > incorrect port 0 in reply from rtp proxy
> > What is the problem here? Also number of file
> > descriptors that RTPProxy can use are set to a million.
> >
> >
> >
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>  --
>  Daniel-Constantin Mierla
>  http://twitter.com/#!/miconda
>   -
> http://www.linkedin.com/in/miconda
>  
>  Kamailio Advanced Training, Berlin, Nov 28-30, 2016 -
> http://www.asipto.com
> 
>  ___ SIP
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> users
>   users>
> 
> 
> >>> --
> >>> Daniel-Constantin Mierla
> >>> http://twitter.com/#!/miconda 
> - http://www.linkedin.com/in/miconda
> >>> 
> >>>

Re: [SR-Users] RTPProxy benchmark

[Dragos Oancea]

Hi

Increase fs.file-max in your /etc/sysctl.conf .
eg: fs.file-max = 5
And then do sysctl -p
Decrease SILENT_TIMEOUT in your rtpengine.conf (eg:SILENT_TIMEOUT=120) -
it's default 1 hour and if some calls don't have media then rtpengine
will just keep the UDP ports in use until this timeout expires.

Regards,
Dragos


On 15/11/2016 09:01, Daniel-Constantin Mierla wrote:
> Are you ending the rtp proxy sessions when the call are ended? What
> rtpproxy functions are you using in the configuration file?
> 
> Cheers,
> Daniel
> 
> 
> On 14/11/16 18:18, Gholamreza Sabery wrote:
>> No not the first time. But over time. I rebooted my system and error
>> is gone! It seems that it happens over time.
>>
>> On Mon, Nov 14, 2016 at 6:12 PM, Daniel-Constantin Mierla
>> mailto:mico...@gmail.com>> wrote:
>>
>> Are you getting the error first time when you reach first 1900
>> sessions? Or after a while, after some previous sessions are ended?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 14/11/16 11:19, Gholamreza Sabery wrote:
>>> I already set these parameters:
>>>
>>> rtpproxy -m 5000 -M 65000
>>>
>>> As well limits for number of open files are set to 100
>>> (ulimit -n). When I increased log level of RTPProxy I saw:
>>> ERR:create_twinlistener:GENERAL: can't create IPv4 socket: Too
>>> many open files in system
>>>
>>>
>>> On Mon, Nov 14, 2016 at 1:46 PM, Daniel-Constantin Mierla
>>> mailto:mico...@gmail.com>> wrote:
>>>
>>> Hello,
>>>
>>> first thing to look at is the port range. There are some
>>> parameter that you can provide to rtpproxy in command line in
>>> order to increase the range of port it can use -- see
>>> 'rtpproxy -h' or 'man rtpproxy'.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 14/11/16 11:14, Gholamreza Sabery wrote:
 Dear Daniel:

 I used a single RTPProxy instance. RTPProxy version =
 20040107. And yes there was traffic for all calls but
 traffic is one-way. One leg sends the call and the other
 just receives it.

 On Mon, Nov 14, 2016 at 1:40 PM, Daniel-Constantin Mierla
 mailto:mico...@gmail.com>> wrote:

 Hello,

 have you used a single rtpproxy instance? Was there RTP
 traffic for all 1900 calls? Is this with rtpproxy 1.2 or
 2.0?

 Cheers,
 Daniel


 On 14/11/16 10:44, Gholamreza Sabery wrote:
> I managed to create about 1900 concurrent calls using a
> single Kamailio and RTPProxy server. But after this
> number RTPProxy returns 0 and the following error is
> shown in the Kamailio log files:
>
> incorrect port 0 in reply from rtp proxy
> What is the problem here? Also number of file
> descriptors that RTPProxy can use are set to a million.
>
>
>
> ___
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> mailing list
> sr-users@lists.sip-router.org
> 
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 
> 

 -- 
 Daniel-Constantin Mierla
 http://twitter.com/#!/miconda
  - 
 http://www.linkedin.com/in/miconda
 
 Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - 
 http://www.asipto.com

 ___ SIP
 Express Router (SER) and Kamailio (OpenSER) - sr-users
 mailing list sr-users@lists.sip-router.org
 
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>>> -- 
>>> Daniel-Constantin Mierla
>>> http://twitter.com/#!/miconda  - 
>>> http://www.linkedin.com/in/miconda
>>> 
>>> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - 
>>> http://www.asipto.com
>>>
>> -- 
>> Daniel-Constantin Mierla
>> http://twitter.com/#!/miconda  - 
>> http://www.linkedin.com/in/miconda
>> 
>> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - 
>> http://www.asipto.com
>>
> -- 
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/i

Re: [SR-Users] RTPProxy benchmark

[Daniel-Constantin Mierla]

Are you ending the rtp proxy sessions when the call are ended? What
rtpproxy functions are you using in the configuration file?

Cheers,
Daniel


On 14/11/16 18:18, Gholamreza Sabery wrote:
> No not the first time. But over time. I rebooted my system and error
> is gone! It seems that it happens over time.
>
> On Mon, Nov 14, 2016 at 6:12 PM, Daniel-Constantin Mierla
> mailto:mico...@gmail.com>> wrote:
>
> Are you getting the error first time when you reach first 1900
> sessions? Or after a while, after some previous sessions are ended?
>
> Cheers,
> Daniel
>
>
> On 14/11/16 11:19, Gholamreza Sabery wrote:
>> I already set these parameters:
>>
>> rtpproxy -m 5000 -M 65000
>>
>> As well limits for number of open files are set to 100
>> (ulimit -n). When I increased log level of RTPProxy I saw:
>> ERR:create_twinlistener:GENERAL: can't create IPv4 socket: Too
>> many open files in system
>>
>>
>> On Mon, Nov 14, 2016 at 1:46 PM, Daniel-Constantin Mierla
>> mailto:mico...@gmail.com>> wrote:
>>
>> Hello,
>>
>> first thing to look at is the port range. There are some
>> parameter that you can provide to rtpproxy in command line in
>> order to increase the range of port it can use -- see
>> 'rtpproxy -h' or 'man rtpproxy'.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 14/11/16 11:14, Gholamreza Sabery wrote:
>>> Dear Daniel:
>>>
>>> I used a single RTPProxy instance. RTPProxy version =
>>> 20040107. And yes there was traffic for all calls but
>>> traffic is one-way. One leg sends the call and the other
>>> just receives it.
>>>
>>> On Mon, Nov 14, 2016 at 1:40 PM, Daniel-Constantin Mierla
>>> mailto:mico...@gmail.com>> wrote:
>>>
>>> Hello,
>>>
>>> have you used a single rtpproxy instance? Was there RTP
>>> traffic for all 1900 calls? Is this with rtpproxy 1.2 or
>>> 2.0?
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 14/11/16 10:44, Gholamreza Sabery wrote:
 I managed to create about 1900 concurrent calls using a
 single Kamailio and RTPProxy server. But after this
 number RTPProxy returns 0 and the following error is
 shown in the Kamailio log files:

 incorrect port 0 in reply from rtp proxy
 What is the problem here? Also number of file
 descriptors that RTPProxy can use are set to a million.



 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users 
 mailing list
 sr-users@lists.sip-router.org
 
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
 
>>>
>>> -- 
>>> Daniel-Constantin Mierla
>>> http://twitter.com/#!/miconda
>>>  - 
>>> http://www.linkedin.com/in/miconda
>>> 
>>> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - 
>>> http://www.asipto.com
>>>
>>> ___ SIP
>>> Express Router (SER) and Kamailio (OpenSER) - sr-users
>>> mailing list sr-users@lists.sip-router.org
>>> 
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>> 
>>>
>>>
>> -- 
>> Daniel-Constantin Mierla
>> http://twitter.com/#!/miconda  - 
>> http://www.linkedin.com/in/miconda
>> 
>> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - 
>> http://www.asipto.com
>>
> -- 
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda  - 
> http://www.linkedin.com/in/miconda
> 
> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - 
> http://www.asipto.com
>
-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
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Re: [SR-Users] RTPProxy benchmark

[Gholamreza Sabery]

No not the first time. But over time. I rebooted my system and error is
gone! It seems that it happens over time.

On Mon, Nov 14, 2016 at 6:12 PM, Daniel-Constantin Mierla  wrote:

> Are you getting the error first time when you reach first 1900 sessions?
> Or after a while, after some previous sessions are ended?
> Cheers,
> Daniel
>
>
> On 14/11/16 11:19, Gholamreza Sabery wrote:
>
> I already set these parameters:
>
> rtpproxy -m 5000 -M 65000
>
> As well limits for number of open files are set to 100 (ulimit -n).
> When I increased log level of RTPProxy I saw:
> ERR:create_twinlistener:GENERAL: can't create IPv4 socket: Too many open
> files in system
>
>
> On Mon, Nov 14, 2016 at 1:46 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> Hello,
>>
>> first thing to look at is the port range. There are some parameter that
>> you can provide to rtpproxy in command line in order to increase the range
>> of port it can use -- see 'rtpproxy -h' or 'man rtpproxy'.
>>
>> Cheers,
>> Daniel
>>
>> On 14/11/16 11:14, Gholamreza Sabery wrote:
>>
>> Dear Daniel:
>>
>> I used a single RTPProxy instance. RTPProxy version = 20040107. And yes
>> there was traffic for all calls but traffic is one-way. One leg sends the
>> call and the other just receives it.
>>
>> On Mon, Nov 14, 2016 at 1:40 PM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> have you used a single rtpproxy instance? Was there RTP traffic for all
>>> 1900 calls? Is this with rtpproxy 1.2 or 2.0?
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 14/11/16 10:44, Gholamreza Sabery wrote:
>>>
>>> I managed to create about 1900 concurrent calls using a single Kamailio
>>> and RTPProxy server. But after this number RTPProxy returns 0 and the
>>> following error is shown in the Kamailio log files:
>>>
>>> incorrect port 0 in reply from rtp proxy
>>>
>>> What is the problem here? Also number of file descriptors that RTPProxy can 
>>> use are set to a million.
>>>
>>>
>>>
>>>
>>> ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>> --
>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
>>> http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
>>>
>>> ___ SIP Express Router
>>> (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users@lists.sip-router.org http://lists.sip-router.org/cg
>>> i-bin/mailman/listinfo/sr-users
>>
>> --
>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
>> http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
>>
>> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
>
>
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Re: [SR-Users] RTPProxy benchmark

[Daniel-Constantin Mierla]

Are you getting the error first time when you reach first 1900 sessions?
Or after a while, after some previous sessions are ended?

Cheers,
Daniel

On 14/11/16 11:19, Gholamreza Sabery wrote:
> I already set these parameters:
>
> rtpproxy -m 5000 -M 65000
>
> As well limits for number of open files are set to 100 (ulimit
> -n). When I increased log level of RTPProxy I saw:
> ERR:create_twinlistener:GENERAL: can't create IPv4 socket: Too many
> open files in system
>
>
> On Mon, Nov 14, 2016 at 1:46 PM, Daniel-Constantin Mierla
> mailto:mico...@gmail.com>> wrote:
>
> Hello,
>
> first thing to look at is the port range. There are some parameter
> that you can provide to rtpproxy in command line in order to
> increase the range of port it can use -- see 'rtpproxy -h' or 'man
> rtpproxy'.
>
> Cheers,
> Daniel
>
>
> On 14/11/16 11:14, Gholamreza Sabery wrote:
>> Dear Daniel:
>>
>> I used a single RTPProxy instance. RTPProxy version = 20040107.
>> And yes there was traffic for all calls but traffic is one-way.
>> One leg sends the call and the other just receives it.
>>
>> On Mon, Nov 14, 2016 at 1:40 PM, Daniel-Constantin Mierla
>> mailto:mico...@gmail.com>> wrote:
>>
>> Hello,
>>
>> have you used a single rtpproxy instance? Was there RTP
>> traffic for all 1900 calls? Is this with rtpproxy 1.2 or 2.0?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 14/11/16 10:44, Gholamreza Sabery wrote:
>>> I managed to create about 1900 concurrent calls using a
>>> single Kamailio and RTPProxy server. But after this number
>>> RTPProxy returns 0 and the following error is shown in the
>>> Kamailio log files:
>>>
>>> incorrect port 0 in reply from rtp proxy
>>> What is the problem here? Also number of file descriptors
>>> that RTPProxy can use are set to a million.
>>>
>>>
>>>
>>> ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>>> list
>>> sr-users@lists.sip-router.org
>>> 
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>> 
>>
>> -- 
>> Daniel-Constantin Mierla
>> http://twitter.com/#!/miconda  - 
>> http://www.linkedin.com/in/miconda
>> 
>> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - 
>> http://www.asipto.com
>>
>> ___ SIP Express
>> Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> 
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>  
>>
> -- 
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda  - 
> http://www.linkedin.com/in/miconda
> 
> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - 
> http://www.asipto.com
>
-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
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Re: [SR-Users] RTPProxy benchmark

[Gholamreza Sabery]

I already set these parameters:

rtpproxy -m 5000 -M 65000

As well limits for number of open files are set to 100 (ulimit -n).
When I increased log level of RTPProxy I saw:
ERR:create_twinlistener:GENERAL: can't create IPv4 socket: Too many open
files in system


On Mon, Nov 14, 2016 at 1:46 PM, Daniel-Constantin Mierla  wrote:

> Hello,
>
> first thing to look at is the port range. There are some parameter that
> you can provide to rtpproxy in command line in order to increase the range
> of port it can use -- see 'rtpproxy -h' or 'man rtpproxy'.
>
> Cheers,
> Daniel
>
> On 14/11/16 11:14, Gholamreza Sabery wrote:
>
> Dear Daniel:
>
> I used a single RTPProxy instance. RTPProxy version = 20040107. And yes
> there was traffic for all calls but traffic is one-way. One leg sends the
> call and the other just receives it.
>
> On Mon, Nov 14, 2016 at 1:40 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> Hello,
>>
>> have you used a single rtpproxy instance? Was there RTP traffic for all
>> 1900 calls? Is this with rtpproxy 1.2 or 2.0?
>>
>> Cheers,
>> Daniel
>>
>> On 14/11/16 10:44, Gholamreza Sabery wrote:
>>
>> I managed to create about 1900 concurrent calls using a single Kamailio
>> and RTPProxy server. But after this number RTPProxy returns 0 and the
>> following error is shown in the Kamailio log files:
>>
>> incorrect port 0 in reply from rtp proxy
>>
>> What is the problem here? Also number of file descriptors that RTPProxy can 
>> use are set to a million.
>>
>>
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>> --
>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
>> http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
>>
>> ___ SIP Express Router (SER)
>> and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org http://lists.sip-router.org/cg
>> i-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
>
>
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Re: [SR-Users] RTPProxy benchmark

[Daniel-Constantin Mierla]

Hello,

first thing to look at is the port range. There are some parameter that
you can provide to rtpproxy in command line in order to increase the
range of port it can use -- see 'rtpproxy -h' or 'man rtpproxy'.

Cheers,
Daniel


On 14/11/16 11:14, Gholamreza Sabery wrote:
> Dear Daniel:
>
> I used a single RTPProxy instance. RTPProxy version = 20040107. And
> yes there was traffic for all calls but traffic is one-way. One leg
> sends the call and the other just receives it.
>
> On Mon, Nov 14, 2016 at 1:40 PM, Daniel-Constantin Mierla
> mailto:mico...@gmail.com>> wrote:
>
> Hello,
>
> have you used a single rtpproxy instance? Was there RTP traffic
> for all 1900 calls? Is this with rtpproxy 1.2 or 2.0?
>
> Cheers,
> Daniel
>
>
> On 14/11/16 10:44, Gholamreza Sabery wrote:
>> I managed to create about 1900 concurrent calls using a single
>> Kamailio and RTPProxy server. But after this number RTPProxy
>> returns 0 and the following error is shown in the Kamailio log files:
>>
>> incorrect port 0 in reply from rtp proxy
>> What is the problem here? Also number of file descriptors that
>> RTPProxy can use are set to a million.
>>
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org 
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> 
>
> -- 
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda  - 
> http://www.linkedin.com/in/miconda
> 
> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - 
> http://www.asipto.com
>
> ___ SIP Express Router
> (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> 
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>  
>
-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
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Re: [SR-Users] RTPProxy benchmark

[Gholamreza Sabery]

Dear Daniel:

I used a single RTPProxy instance. RTPProxy version = 20040107. And yes
there was traffic for all calls but traffic is one-way. One leg sends the
call and the other just receives it.

On Mon, Nov 14, 2016 at 1:40 PM, Daniel-Constantin Mierla  wrote:

> Hello,
>
> have you used a single rtpproxy instance? Was there RTP traffic for all
> 1900 calls? Is this with rtpproxy 1.2 or 2.0?
>
> Cheers,
> Daniel
>
> On 14/11/16 10:44, Gholamreza Sabery wrote:
>
> I managed to create about 1900 concurrent calls using a single Kamailio
> and RTPProxy server. But after this number RTPProxy returns 0 and the
> following error is shown in the Kamailio log files:
>
> incorrect port 0 in reply from rtp proxy
>
> What is the problem here? Also number of file descriptors that RTPProxy can 
> use are set to a million.
>
>
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
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>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
>
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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>
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Re: [SR-Users] RTPProxy benchmark

[Daniel-Constantin Mierla]

Hello,

have you used a single rtpproxy instance? Was there RTP traffic for all
1900 calls? Is this with rtpproxy 1.2 or 2.0?

Cheers,
Daniel


On 14/11/16 10:44, Gholamreza Sabery wrote:
> I managed to create about 1900 concurrent calls using a single
> Kamailio and RTPProxy server. But after this number RTPProxy returns 0
> and the following error is shown in the Kamailio log files:
>
> incorrect port 0 in reply from rtp proxy
> What is the problem here? Also number of file descriptors that
> RTPProxy can use are set to a million.
>
>
>
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-- 
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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

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Re: [SR-Users] RTPProxy

[Daniel-Constantin Mierla]

Hello,

thanks for all those details, very useful ...

To be clear -- the issue of using high cpu on idle (no active calls) was
with rtpproxy v1.2 on a centos (iirc, v6), not with rtpproxy 2.0. On
debian, same version of rtpproxy was not exposing this. I was just
curios to see if anyone else saw it ... might have been just that system...

Cheers,
Daniel


On 19/10/16 20:00, Maxim Sobolev wrote:
> Just a little comment on the numbers that I've thrown out earlier
> today. Those are probably somewhat pessimistic, with some creative
> tuneup you can probably go much higher. But we also constrained by
> some other considerations (i.e. running fully redundant network
> connection with FEC, full firewall etc, custom OS), so those are what
> we get.
>
> Also, I wanted to point to the list, speaking about number of sessions
> is pretty much pointless, as the main thing that keeps us busy is
> packet per second rate. Since same 10,000 sessions might translate to
> as much as half of PPS rate if use 10ms ptype versus 20ms ptype. Our
> limit at this point of time is some 450k PPS in and 450k PPS out, 16
> cores, FreeBSD 10.3, which could be either 4.5k sessions with 10ms
> packets or 9k sessions with 20ms or somewhere in between if you have
> mixed traffic (as most of our customers do). Latest Linux kernels
> might get better contention control on higher CPU count systems, or at
> least it is what I've seen on some of the benchmarks not so long time
> ago, We've planned to run some evaluations but have not got time to do
> so yet.
>
> On top of that, even if you can push say 1 million PPS through single
> tuned up box (10k sessions at 10ms), some other constrains may arise.
> Most of the general-purpose DC providers we've encountered in our
> somewhat limited practice, design their networks with much lower PPS
> per port in mind. It's often an issue with a new DC here that we bump
> into all sorts of automated DDoS prevention systems once we reach
> 100-200k PPS per box/port. So at the end of the day it might be more
> practical and economical to run bunch of the smaller nodes and spread
> the load across them using something like rtp_cluster rather than try
> to cram all that traffic into a single box/port.
>
> -Max

-- 
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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

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Re: [SR-Users] RTPProxy

[Alex Balashov]

My personal opinion on this is that it should be very low-priority. It's 
one of those problems that takes 99% effort for 1% marginal results, and 
even then, rather imperfect ones.


For almost any service provider, having a media relay while calls are on 
it is not the worst possible problem—certainly not when set against the 
expense and complexity of solving it. If the calls are properly 
distributed across an array of media servers, it shouldn't be a total 
service-impacting event for anyone, anyway.


On 10/19/2016 10:54 AM, Maxim Sobolev wrote:


Arsen, there is no readily-made solution with rtpproxy unfortunately for
that. Some time around 1.0 times circa 2007-2009, somebody submitted a
very rough patch to implement master/hot-standby scenario, but the patch
was not production-ready back then and the contributor was not available
to refine it further, so it ended up stashed somewhere on the branch in
git. I'd be happy to say we are working on that, but our resources are
limited and priority is to get features like SRTP, trans-coding and
video support. As always, pull requests/patches are welcome. With 2.x
code is much more modular, so it should be easier to get something like
that working.

-Max

On Wed, Oct 19, 2016 at 3:55 AM, Arsen mailto:arsen.semio...@gmail.com>> wrote:

Hi guys,

In addition to this interesting and useful thread, what is the best
way to implement media session recovery, for example in
Active/Passive HA scenario?
I know that it is possible with rtpengine (redis db), is it possible
with rtpproxy?



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

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Re: [SR-Users] RTPProxy

[Maxim Sobolev]

Just a little comment on the numbers that I've thrown out earlier today.
Those are probably somewhat pessimistic, with some creative tuneup you can
probably go much higher. But we also constrained by some other
considerations (i.e. running fully redundant network connection with FEC,
full firewall etc, custom OS), so those are what we get.

Also, I wanted to point to the list, speaking about number of sessions is
pretty much pointless, as the main thing that keeps us busy is packet per
second rate. Since same 10,000 sessions might translate to as much as half
of PPS rate if use 10ms ptype versus 20ms ptype. Our limit at this point of
time is some 450k PPS in and 450k PPS out, 16 cores, FreeBSD 10.3, which
could be either 4.5k sessions with 10ms packets or 9k sessions with 20ms or
somewhere in between if you have mixed traffic (as most of our customers
do). Latest Linux kernels might get better contention control on higher CPU
count systems, or at least it is what I've seen on some of the benchmarks
not so long time ago, We've planned to run some evaluations but have not
got time to do so yet.

On top of that, even if you can push say 1 million PPS through single tuned
up box (10k sessions at 10ms), some other constrains may arise. Most of the
general-purpose DC providers we've encountered in our somewhat limited
practice, design their networks with much lower PPS per port in mind. It's
often an issue with a new DC here that we bump into all sorts of automated
DDoS prevention systems once we reach 100-200k PPS per box/port. So at the
end of the day it might be more practical and economical to run bunch of
the smaller nodes and spread the load across them using something like
rtp_cluster rather than try to cram all that traffic into a single box/port.

-Max
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Re: [SR-Users] RTPProxy

[Maxim Sobolev]

Arsen, there is no readily-made solution with rtpproxy unfortunately for
that. Some time around 1.0 times circa 2007-2009, somebody submitted a very
rough patch to implement master/hot-standby scenario, but the patch was not
production-ready back then and the contributor was not available to refine
it further, so it ended up stashed somewhere on the branch in git. I'd be
happy to say we are working on that, but our resources are limited and
priority is to get features like SRTP, trans-coding and video support. As
always, pull requests/patches are welcome. With 2.x code is much more
modular, so it should be easier to get something like that working.

-Max

On Wed, Oct 19, 2016 at 3:55 AM, Arsen  wrote:

> Hi guys,
>
> In addition to this interesting and useful thread, what is the best way to
> implement media session recovery, for example in Active/Passive HA scenario?
> I know that it is possible with rtpengine (redis db), is it possible with
> rtpproxy?
>
>
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Re: [SR-Users] RTPProxy

[Maxim Sobolev]

Daniel, thank you for your interest. Yes, there were many architectural
changes between 1.x and 2.0. The most noticeable is that we've decoupled
I/O from the control channel handling and also split I/O into two threads,
one for poll/receive and the second one for the sending. We've also
refactored our scheduling algorithms to provide much better performance
jitter-wise. There are more threads than two in 2.0, but most of them are
not using much CPU if any. As such as a rule of thumb you'd want to run max
one rtpproxy per two CPU cores with 2.0, up from one instance per core with
1.x. Yes, 1000 bi-directional streams is realistic value, at least in a
situation when you have multiple rtpproxy instances running concurrently on
the box which creates non-trivial overhead in terms of system calls
contention. You can probably go higher than 1,000 if you have single
instance and/or latest CPU. There is detailed 2.0 release notes document
available at our github project.

In terms of cpu usage while idle, I am not aware of any major issues with
2.0. That being said, rtpproxy is known to be somewhat wasteful in general
while running with no load, it still does 200 poll's a second and that
tends to consume <1% of single core CPU per running instance. I've got some
ideas on how to fix it, but that would not be available until 2.2 is out
somewhere in 2017. We are working on getting 2.1 out, which would include
new plain text accounting module more refinements and improvements. Among
those, we've added experimental support for running control channel over
TCPv4 and TCPv6 sockets, still something that needs to be integrated back
into all children of SER.

We are aware that packaging is somewhat behind in some distros*. However,
we ourselves and most of our biggest customers roll their own builds
anyway, so we just rely on the community to take care of. That being said,
we always welcome pull requests to get stuff that might help others to
package it on FooLinux.

Please don't hesitate to ask if something is unclear or if you have more
questions. Thanks!

-Max
*) Sometimes I get a feeling that there are too many to chose from these
days, but oh well... :)

On Wed, Oct 19, 2016 at 1:19 AM, Daniel-Constantin Mierla  wrote:

> Hello Maxim,
> given the discussion here, I would like to get some updates for myself
> regarding 2.0 in terms of capacity and other stuff.
>
> I was using rtpproxy 1.x with kamailio doing load balancing across many
> instances of rtpproxy. I was using 1000 streams as estimation for one
> instance and I see it's what you mentioned as well. Is it the recommended
> (or the good) value for 2.0? Most of deployments still use v1.2, given it's
> presence in stable/old OS distros.
>
> It's any relevant architectural change in 2.0? Like more threads used by
> the app or other I/O refactoring? Iirc, v1.x uses one for control commands?
>
> I wanted to report at some point, with v1.x, on some centos (iirc), when
> there was no active call, rtpproxy was eating a lot of cpu. With a call (or
> more) going on, the cpu went to normal. I think it was like waiting for I/O
> was using the cpu. Switching to debian was a solution at that moment, so
> might not be rtpproxy, but I am wondering if you or anyone else faced same
> issue. Also, if I am not wrong, the person that reported to me said that
> 2.0 didn't revealed the same behaviour.
>
> Cheers,
> Daniel
>
>
> On 19/10/16 09:46, Maxim Sobolev wrote:
>
> Alex, no problem. Nobody knows everything. :)
>
> -Max
>
> On Wed, Oct 19, 2016 at 12:35 AM, Alex Balashov  > wrote:
>
>> Hi Maxim,
>>
>> Duly noted! I certainly did not intend to mislead anyone or to be
>> disingenuous; I gave information that was, to the best of my knowledge,
>> true. I appreciate your followup and clarification, which certainly is
>> useful for my own knowledge as well!
>>
>> My sincere apologies...
>>
>> -- Alex
>>
>>
>> On October 19, 2016 3:32:24 AM EDT, Maxim Sobolev 
>> wrote:
>> >Alex, with all due respect, things you said about rtpproxy capacity is
>> >somewhat outdated and misleading. We have some nodes in the field, that
>> >handle 5,000-6,000 rtp sessions in peak. Those are running 6 rtpproxy
>> >instances, 1,000 sessions each.  2-3 year old CPUs, 12 cores in total.
>> >
>> >We also have an open source solution called rtp_cluster, which allows
>> >building larger scale deployments, for at least up to 50,000
>> >bidirectional
>> >streams using multiple nodes running rtpproxy. Available here
>> >https://github.com/sippy/rtp_cluster. You are also welcome to check our
>> >talk last summer at the opensips devsummit in Austin where we gave it
>> >some
>> >limelight.
>> >
>> >So you are off by two orders of magnitude roughly with regards to the
>> >capacity. :)
>> >
>> >And yes, we've been happily running large deployments at AWS for at
>> >least 6
>> >years now.
>> >
>> >Rodrigo, speaking about your original question, I could not tell much
>> >about
>> >rtpengine due to a lack of practical exper

Re: [SR-Users] RTPProxy

[Arsen]

Hi guys,

In addition to this interesting and useful thread, what is the best way to
implement media session recovery, for example in Active/Passive HA scenario?
I know that it is possible with rtpengine (redis db), is it possible with
rtpproxy?

Thanks,
Arsen.

On Wed, Oct 19, 2016 at 11:19 AM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

> Hello Maxim,
> given the discussion here, I would like to get some updates for myself
> regarding 2.0 in terms of capacity and other stuff.
>
> I was using rtpproxy 1.x with kamailio doing load balancing across many
> instances of rtpproxy. I was using 1000 streams as estimation for one
> instance and I see it's what you mentioned as well. Is it the recommended
> (or the good) value for 2.0? Most of deployments still use v1.2, given it's
> presence in stable/old OS distros.
>
> It's any relevant architectural change in 2.0? Like more threads used by
> the app or other I/O refactoring? Iirc, v1.x uses one for control commands?
>
> I wanted to report at some point, with v1.x, on some centos (iirc), when
> there was no active call, rtpproxy was eating a lot of cpu. With a call (or
> more) going on, the cpu went to normal. I think it was like waiting for I/O
> was using the cpu. Switching to debian was a solution at that moment, so
> might not be rtpproxy, but I am wondering if you or anyone else faced same
> issue. Also, if I am not wrong, the person that reported to me said that
> 2.0 didn't revealed the same behaviour.
>
> Cheers,
> Daniel
>
>
> On 19/10/16 09:46, Maxim Sobolev wrote:
>
> Alex, no problem. Nobody knows everything. :)
>
> -Max
>
> On Wed, Oct 19, 2016 at 12:35 AM, Alex Balashov  > wrote:
>
>> Hi Maxim,
>>
>> Duly noted! I certainly did not intend to mislead anyone or to be
>> disingenuous; I gave information that was, to the best of my knowledge,
>> true. I appreciate your followup and clarification, which certainly is
>> useful for my own knowledge as well!
>>
>> My sincere apologies...
>>
>> -- Alex
>>
>>
>> On October 19, 2016 3:32:24 AM EDT, Maxim Sobolev 
>> wrote:
>> >Alex, with all due respect, things you said about rtpproxy capacity is
>> >somewhat outdated and misleading. We have some nodes in the field, that
>> >handle 5,000-6,000 rtp sessions in peak. Those are running 6 rtpproxy
>> >instances, 1,000 sessions each.  2-3 year old CPUs, 12 cores in total.
>> >
>> >We also have an open source solution called rtp_cluster, which allows
>> >building larger scale deployments, for at least up to 50,000
>> >bidirectional
>> >streams using multiple nodes running rtpproxy. Available here
>> >https://github.com/sippy/rtp_cluster. You are also welcome to check our
>> >talk last summer at the opensips devsummit in Austin where we gave it
>> >some
>> >limelight.
>> >
>> >So you are off by two orders of magnitude roughly with regards to the
>> >capacity. :)
>> >
>> >And yes, we've been happily running large deployments at AWS for at
>> >least 6
>> >years now.
>> >
>> >Rodrigo, speaking about your original question, I could not tell much
>> >about
>> >rtpengine due to a lack of practical experience with it. But from what
>> >I
>> >read on its website it seems to be logical continuation of the
>> >mediaproxy
>> >package packed with some cutting edge sexy features.
>> >
>> >In a nutshell rtpproxy and mediaproxy/rtpengine are just two
>> >independently
>> >developed pieces of software, doing somewhat similar function. What
>> >would
>> >work in your particular setting depends on your requirements and
>> >constraints.
>> >
>> >Here at Sippy Labs we focus on stability, compatibility and portability
>> >for
>> >a predominantly regular audio traffic.
>> >
>> >We also have a test suite that check compatibility of the latest
>> >production
>> >and development versions of the rtpproxy against array of different SIP
>> >engines, including Kamailio. https://travis-ci.org/sippy/voiptests
>> >
>> >So with rtpproxy you are not locked in into single SIP engine, you can
>> >mix
>> >and match to fit your particular goal.
>> >
>> >And yes, last but not least, all our code is BSD licensed, so you can
>> >build
>> >you proprietary box that uses it.
>> >
>> >Hope it helps.
>> >
>> >-Max
>> >
>> >On Oct 17, 2016 11:33 AM, "Alex Balashov" 
>> >wrote:
>> >
>> >> On 10/17/2016 02:29 PM, Rodrigo Moreira wrote:
>> >>
>> >> What is difference between modules rtpproxy and rtpengine?
>> >>>
>> >>
>> >> rtpproxy is a userspace process which, historically, has a relatively
>> >> limited call throughput capacity (maybe a few hundred calls), though
>> >this
>> >> might be addressed to some degree in rtpproxy 2.0. Nevertheless, it
>> >has
>> >> been commonly used and well supported in the *SER family for long
>> >time.
>> >>
>> >> RTPEngine is a newer initiative from Sipwise, and uses kernel-mode
>> >> forwarding to achieve close to on-the-wire RTP forwarding speeds. It
>> >can do
>> >> 10,000+ concurrent bidirectional RTP streams. It also has lots of
>> >other
>> >> features which can 

Re: [SR-Users] RTPProxy

[Daniel-Constantin Mierla]

Hello Maxim,

given the discussion here, I would like to get some updates for myself
regarding 2.0 in terms of capacity and other stuff.

I was using rtpproxy 1.x with kamailio doing load balancing across many
instances of rtpproxy. I was using 1000 streams as estimation for one
instance and I see it's what you mentioned as well. Is it the
recommended (or the good) value for 2.0? Most of deployments still use
v1.2, given it's presence in stable/old OS distros.

It's any relevant architectural change in 2.0? Like more threads used by
the app or other I/O refactoring? Iirc, v1.x uses one for control commands?

I wanted to report at some point, with v1.x, on some centos (iirc), when
there was no active call, rtpproxy was eating a lot of cpu. With a call
(or more) going on, the cpu went to normal. I think it was like waiting
for I/O was using the cpu. Switching to debian was a solution at that
moment, so might not be rtpproxy, but I am wondering if you or anyone
else faced same issue. Also, if I am not wrong, the person that reported
to me said that 2.0 didn't revealed the same behaviour.

Cheers,
Daniel

On 19/10/16 09:46, Maxim Sobolev wrote:
> Alex, no problem. Nobody knows everything. :) 
>
> -Max
>
> On Wed, Oct 19, 2016 at 12:35 AM, Alex Balashov
> mailto:abalas...@evaristesys.com>> wrote:
>
> Hi Maxim,
>
> Duly noted! I certainly did not intend to mislead anyone or to be
> disingenuous; I gave information that was, to the best of my
> knowledge, true. I appreciate your followup and clarification,
> which certainly is useful for my own knowledge as well!
>
> My sincere apologies...
>
> -- Alex
>
>
> On October 19, 2016 3:32:24 AM EDT, Maxim Sobolev
> mailto:sobo...@sippysoft.com>> wrote:
> >Alex, with all due respect, things you said about rtpproxy
> capacity is
> >somewhat outdated and misleading. We have some nodes in the
> field, that
> >handle 5,000-6,000 rtp sessions in peak. Those are running 6 rtpproxy
> >instances, 1,000 sessions each.  2-3 year old CPUs, 12 cores in
> total.
> >
> >We also have an open source solution called rtp_cluster, which allows
> >building larger scale deployments, for at least up to 50,000
> >bidirectional
> >streams using multiple nodes running rtpproxy. Available here
> >https://github.com/sippy/rtp_cluster
> . You are also welcome to
> check our
> >talk last summer at the opensips devsummit in Austin where we gave it
> >some
> >limelight.
> >
> >So you are off by two orders of magnitude roughly with regards to the
> >capacity. :)
> >
> >And yes, we've been happily running large deployments at AWS for at
> >least 6
> >years now.
> >
> >Rodrigo, speaking about your original question, I could not tell much
> >about
> >rtpengine due to a lack of practical experience with it. But from
> what
> >I
> >read on its website it seems to be logical continuation of the
> >mediaproxy
> >package packed with some cutting edge sexy features.
> >
> >In a nutshell rtpproxy and mediaproxy/rtpengine are just two
> >independently
> >developed pieces of software, doing somewhat similar function. What
> >would
> >work in your particular setting depends on your requirements and
> >constraints.
> >
> >Here at Sippy Labs we focus on stability, compatibility and
> portability
> >for
> >a predominantly regular audio traffic.
> >
> >We also have a test suite that check compatibility of the latest
> >production
> >and development versions of the rtpproxy against array of
> different SIP
> >engines, including Kamailio.
> https://travis-ci.org/sippy/voiptests
> 
> >
> >So with rtpproxy you are not locked in into single SIP engine,
> you can
> >mix
> >and match to fit your particular goal.
> >
> >And yes, last but not least, all our code is BSD licensed, so you can
> >build
> >you proprietary box that uses it.
> >
> >Hope it helps.
> >
> >-Max
> >
> >On Oct 17, 2016 11:33 AM, "Alex Balashov"
> mailto:abalas...@evaristesys.com>>
> >wrote:
> >
> >> On 10/17/2016 02:29 PM, Rodrigo Moreira wrote:
> >>
> >> What is difference between modules rtpproxy and rtpengine?
> >>>
> >>
> >> rtpproxy is a userspace process which, historically, has a
> relatively
> >> limited call throughput capacity (maybe a few hundred calls),
> though
> >this
> >> might be addressed to some degree in rtpproxy 2.0. Nevertheless, it
> >has
> >> been commonly used and well supported in the *SER family for long
> >time.
> >>
> >> RTPEngine is a newer initiative from Sipwise, and uses kernel-mode
> >> forwarding to achieve close to on-the-wire RTP forwarding
> speeds. It
> 

Re: [SR-Users] RTPProxy

[Alex Balashov]

And yes, I was remiss in failing to mention that an effective solution 
to scaling out rtpproxy is to bind multiple instances with different 
core affinities.


--
Alex Balashov | Principal | Evariste Systems LLC

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

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Re: [SR-Users] RTPProxy

[Maxim Sobolev]

Alex, no problem. Nobody knows everything. :)

-Max

On Wed, Oct 19, 2016 at 12:35 AM, Alex Balashov 
wrote:

> Hi Maxim,
>
> Duly noted! I certainly did not intend to mislead anyone or to be
> disingenuous; I gave information that was, to the best of my knowledge,
> true. I appreciate your followup and clarification, which certainly is
> useful for my own knowledge as well!
>
> My sincere apologies...
>
> -- Alex
>
>
> On October 19, 2016 3:32:24 AM EDT, Maxim Sobolev 
> wrote:
> >Alex, with all due respect, things you said about rtpproxy capacity is
> >somewhat outdated and misleading. We have some nodes in the field, that
> >handle 5,000-6,000 rtp sessions in peak. Those are running 6 rtpproxy
> >instances, 1,000 sessions each.  2-3 year old CPUs, 12 cores in total.
> >
> >We also have an open source solution called rtp_cluster, which allows
> >building larger scale deployments, for at least up to 50,000
> >bidirectional
> >streams using multiple nodes running rtpproxy. Available here
> >https://github.com/sippy/rtp_cluster. You are also welcome to check our
> >talk last summer at the opensips devsummit in Austin where we gave it
> >some
> >limelight.
> >
> >So you are off by two orders of magnitude roughly with regards to the
> >capacity. :)
> >
> >And yes, we've been happily running large deployments at AWS for at
> >least 6
> >years now.
> >
> >Rodrigo, speaking about your original question, I could not tell much
> >about
> >rtpengine due to a lack of practical experience with it. But from what
> >I
> >read on its website it seems to be logical continuation of the
> >mediaproxy
> >package packed with some cutting edge sexy features.
> >
> >In a nutshell rtpproxy and mediaproxy/rtpengine are just two
> >independently
> >developed pieces of software, doing somewhat similar function. What
> >would
> >work in your particular setting depends on your requirements and
> >constraints.
> >
> >Here at Sippy Labs we focus on stability, compatibility and portability
> >for
> >a predominantly regular audio traffic.
> >
> >We also have a test suite that check compatibility of the latest
> >production
> >and development versions of the rtpproxy against array of different SIP
> >engines, including Kamailio. https://travis-ci.org/sippy/voiptests
> >
> >So with rtpproxy you are not locked in into single SIP engine, you can
> >mix
> >and match to fit your particular goal.
> >
> >And yes, last but not least, all our code is BSD licensed, so you can
> >build
> >you proprietary box that uses it.
> >
> >Hope it helps.
> >
> >-Max
> >
> >On Oct 17, 2016 11:33 AM, "Alex Balashov" 
> >wrote:
> >
> >> On 10/17/2016 02:29 PM, Rodrigo Moreira wrote:
> >>
> >> What is difference between modules rtpproxy and rtpengine?
> >>>
> >>
> >> rtpproxy is a userspace process which, historically, has a relatively
> >> limited call throughput capacity (maybe a few hundred calls), though
> >this
> >> might be addressed to some degree in rtpproxy 2.0. Nevertheless, it
> >has
> >> been commonly used and well supported in the *SER family for long
> >time.
> >>
> >> RTPEngine is a newer initiative from Sipwise, and uses kernel-mode
> >> forwarding to achieve close to on-the-wire RTP forwarding speeds. It
> >can do
> >> 10,000+ concurrent bidirectional RTP streams. It also has lots of
> >other
> >> features which can be useful in, for example, running an RTP relay in
> >1:1
> >> NAT environments such as AWS, or in enabling WebRTC.
> >>
> >> However, it is a bit more complicated to set up than vanilla
> >rtpproxy. Not
> >> much more, though.
> >>
> >> -- Alex
> >>
> >> --
> >> Alex Balashov | Principal | Evariste Systems LLC
> >>
> >> Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
> >> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
> >>
> >> ___
> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
> >list
> >> sr-users@lists.sip-router.org
> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >>
> >
> >
> >
> >
> >___
> >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> >sr-users@lists.sip-router.org
> >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> -- Alex
>
> --
> Principal, Evariste Systems LLC (www.evaristesys.com)
>
> Sent from my Google Nexus.
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474
Tel (Toll-Free): +1-855-747-7779
Fax: +1-866-857-6942
Web: http://www.sippysoft.com
MSN: sa...@sippysoft.com
Skype: SippySoft
___
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Re: [SR-Users] RTPProxy

[Alex Balashov]

Hi Maxim, 

Duly noted! I certainly did not intend to mislead anyone or to be disingenuous; 
I gave information that was, to the best of my knowledge, true. I appreciate 
your followup and clarification, which certainly is useful for my own knowledge 
as well! 

My sincere apologies...

-- Alex


On October 19, 2016 3:32:24 AM EDT, Maxim Sobolev  wrote:
>Alex, with all due respect, things you said about rtpproxy capacity is
>somewhat outdated and misleading. We have some nodes in the field, that
>handle 5,000-6,000 rtp sessions in peak. Those are running 6 rtpproxy
>instances, 1,000 sessions each.  2-3 year old CPUs, 12 cores in total.
>
>We also have an open source solution called rtp_cluster, which allows
>building larger scale deployments, for at least up to 50,000
>bidirectional
>streams using multiple nodes running rtpproxy. Available here
>https://github.com/sippy/rtp_cluster. You are also welcome to check our
>talk last summer at the opensips devsummit in Austin where we gave it
>some
>limelight.
>
>So you are off by two orders of magnitude roughly with regards to the
>capacity. :)
>
>And yes, we've been happily running large deployments at AWS for at
>least 6
>years now.
>
>Rodrigo, speaking about your original question, I could not tell much
>about
>rtpengine due to a lack of practical experience with it. But from what
>I
>read on its website it seems to be logical continuation of the
>mediaproxy
>package packed with some cutting edge sexy features.
>
>In a nutshell rtpproxy and mediaproxy/rtpengine are just two
>independently
>developed pieces of software, doing somewhat similar function. What
>would
>work in your particular setting depends on your requirements and
>constraints.
>
>Here at Sippy Labs we focus on stability, compatibility and portability
>for
>a predominantly regular audio traffic.
>
>We also have a test suite that check compatibility of the latest
>production
>and development versions of the rtpproxy against array of different SIP
>engines, including Kamailio. https://travis-ci.org/sippy/voiptests
>
>So with rtpproxy you are not locked in into single SIP engine, you can
>mix
>and match to fit your particular goal.
>
>And yes, last but not least, all our code is BSD licensed, so you can
>build
>you proprietary box that uses it.
>
>Hope it helps.
>
>-Max
>
>On Oct 17, 2016 11:33 AM, "Alex Balashov" 
>wrote:
>
>> On 10/17/2016 02:29 PM, Rodrigo Moreira wrote:
>>
>> What is difference between modules rtpproxy and rtpengine?
>>>
>>
>> rtpproxy is a userspace process which, historically, has a relatively
>> limited call throughput capacity (maybe a few hundred calls), though
>this
>> might be addressed to some degree in rtpproxy 2.0. Nevertheless, it
>has
>> been commonly used and well supported in the *SER family for long
>time.
>>
>> RTPEngine is a newer initiative from Sipwise, and uses kernel-mode
>> forwarding to achieve close to on-the-wire RTP forwarding speeds. It
>can do
>> 10,000+ concurrent bidirectional RTP streams. It also has lots of
>other
>> features which can be useful in, for example, running an RTP relay in
>1:1
>> NAT environments such as AWS, or in enabling WebRTC.
>>
>> However, it is a bit more complicated to set up than vanilla
>rtpproxy. Not
>> much more, though.
>>
>> -- Alex
>>
>> --
>> Alex Balashov | Principal | Evariste Systems LLC
>>
>> Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
>
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-- Alex

--
Principal, Evariste Systems LLC (www.evaristesys.com)

Sent from my Google Nexus.


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Re: [SR-Users] RTPProxy

[Maxim Sobolev]

Alex, with all due respect, things you said about rtpproxy capacity is
somewhat outdated and misleading. We have some nodes in the field, that
handle 5,000-6,000 rtp sessions in peak. Those are running 6 rtpproxy
instances, 1,000 sessions each.  2-3 year old CPUs, 12 cores in total.

We also have an open source solution called rtp_cluster, which allows
building larger scale deployments, for at least up to 50,000 bidirectional
streams using multiple nodes running rtpproxy. Available here
https://github.com/sippy/rtp_cluster. You are also welcome to check our
talk last summer at the opensips devsummit in Austin where we gave it some
limelight.

So you are off by two orders of magnitude roughly with regards to the
capacity. :)

And yes, we've been happily running large deployments at AWS for at least 6
years now.

Rodrigo, speaking about your original question, I could not tell much about
rtpengine due to a lack of practical experience with it. But from what I
read on its website it seems to be logical continuation of the mediaproxy
package packed with some cutting edge sexy features.

In a nutshell rtpproxy and mediaproxy/rtpengine are just two independently
developed pieces of software, doing somewhat similar function. What would
work in your particular setting depends on your requirements and
constraints.

Here at Sippy Labs we focus on stability, compatibility and portability for
a predominantly regular audio traffic.

We also have a test suite that check compatibility of the latest production
and development versions of the rtpproxy against array of different SIP
engines, including Kamailio. https://travis-ci.org/sippy/voiptests

So with rtpproxy you are not locked in into single SIP engine, you can mix
and match to fit your particular goal.

And yes, last but not least, all our code is BSD licensed, so you can build
you proprietary box that uses it.

Hope it helps.

-Max

On Oct 17, 2016 11:33 AM, "Alex Balashov"  wrote:

> On 10/17/2016 02:29 PM, Rodrigo Moreira wrote:
>
> What is difference between modules rtpproxy and rtpengine?
>>
>
> rtpproxy is a userspace process which, historically, has a relatively
> limited call throughput capacity (maybe a few hundred calls), though this
> might be addressed to some degree in rtpproxy 2.0. Nevertheless, it has
> been commonly used and well supported in the *SER family for long time.
>
> RTPEngine is a newer initiative from Sipwise, and uses kernel-mode
> forwarding to achieve close to on-the-wire RTP forwarding speeds. It can do
> 10,000+ concurrent bidirectional RTP streams. It also has lots of other
> features which can be useful in, for example, running an RTP relay in 1:1
> NAT environments such as AWS, or in enabling WebRTC.
>
> However, it is a bit more complicated to set up than vanilla rtpproxy. Not
> much more, though.
>
> -- Alex
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> ___
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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Re: [SR-Users] RTPProxy

[Alex Balashov]

On 10/17/2016 02:29 PM, Rodrigo Moreira wrote:


What is difference between modules rtpproxy and rtpengine?


rtpproxy is a userspace process which, historically, has a relatively 
limited call throughput capacity (maybe a few hundred calls), though 
this might be addressed to some degree in rtpproxy 2.0. Nevertheless, it 
has been commonly used and well supported in the *SER family for long time.


RTPEngine is a newer initiative from Sipwise, and uses kernel-mode 
forwarding to achieve close to on-the-wire RTP forwarding speeds. It can 
do 10,000+ concurrent bidirectional RTP streams. It also has lots of 
other features which can be useful in, for example, running an RTP relay 
in 1:1 NAT environments such as AWS, or in enabling WebRTC.


However, it is a bit more complicated to set up than vanilla rtpproxy. 
Not much more, though.


-- Alex

--
Alex Balashov | Principal | Evariste Systems LLC

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

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Re: [SR-Users] RTPProxy Issues

[TEG AMJG]

Hi again

Yes, i feel very dumb because i actually faced this problem in my testing
with SIPp and with Asterisk before and it was all about open file
description. I probably didnt think about it because I couldnt find
anything in the logs about it which it is quite weird, the only thing that
logs is the one mentioned before, I feel sorry for wasting your time but at
the same time i have to thanks to you guys for helped me out.

Best Regards.

Alejandro

2016-07-17 23:43 GMT-04:00 Maxim Sobolev :

> Yes, that's probably it. There should be also some error in the log that
> rtpproxy emits, so you might want to check that. I see people run into this
> from time to time, perhaps we need to check and put out a big warning in
> red if the OS limit appers to be too low?
>
> -Max
>
> On Sun, Jul 17, 2016 at 3:18 PM, Nathan Angelacos 
> wrote:
>
>> On 07/17/2016 05:49 PM, TEG AMJG wrote:
>>
>>> Dear list
>>>
>>> I am quite new to Kamailio and i have been able to solve some NAT
>>> Traversal issues with symmetric SIP+RTP putting kamailio+rtpproxy behind
>>> NAT, i am also load balancing some asterisk boxes for transcoding and some
>>> other services like voicemail. Also i am using SIPp for load testing
>>>
>>> Now the question is that, while everything is working great with NAT
>>> (even when i am using SIPp for testing with low load) when i am about to
>>> test it with more than 150 calls or something i am starting to get the
>>> following error:
>>>
>>> /kamailio[34549]: ERROR: rtpproxy [rtpproxy.c:2735]: force_rtp_proxy():
>>> incorrect port 0 in reply from rtp proxy/
>>>
>>
>>
>> A guess, but each call takes 4 file descriptors, so you might be running
>> into the normal os limit of 1024 open files per process.
>> Recent rtpproxy versions have the -L option to increase the limit. It
>> should be 4 times the number of concurrent calls you expect to handle.
>>
>> Again, just a guess.
>>
>>
>>
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>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
>
>
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>
>


-- 
-
Saludos a todos
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Re: [SR-Users] RTPProxy Issues

[Maxim Sobolev]

Yes, that's probably it. There should be also some error in the log that
rtpproxy emits, so you might want to check that. I see people run into this
from time to time, perhaps we need to check and put out a big warning in
red if the OS limit appers to be too low?

-Max

On Sun, Jul 17, 2016 at 3:18 PM, Nathan Angelacos 
wrote:

> On 07/17/2016 05:49 PM, TEG AMJG wrote:
>
>> Dear list
>>
>> I am quite new to Kamailio and i have been able to solve some NAT
>> Traversal issues with symmetric SIP+RTP putting kamailio+rtpproxy behind
>> NAT, i am also load balancing some asterisk boxes for transcoding and some
>> other services like voicemail. Also i am using SIPp for load testing
>>
>> Now the question is that, while everything is working great with NAT
>> (even when i am using SIPp for testing with low load) when i am about to
>> test it with more than 150 calls or something i am starting to get the
>> following error:
>>
>> /kamailio[34549]: ERROR: rtpproxy [rtpproxy.c:2735]: force_rtp_proxy():
>> incorrect port 0 in reply from rtp proxy/
>>
>
>
> A guess, but each call takes 4 file descriptors, so you might be running
> into the normal os limit of 1024 open files per process.
> Recent rtpproxy versions have the -L option to increase the limit. It
> should be 4 times the number of concurrent calls you expect to handle.
>
> Again, just a guess.
>
>
>
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Re: [SR-Users] RTPProxy Issues

[Nathan Angelacos]

On 07/17/2016 05:49 PM, TEG AMJG wrote:

Dear list

I am quite new to Kamailio and i have been able to solve some NAT 
Traversal issues with symmetric SIP+RTP putting kamailio+rtpproxy 
behind NAT, i am also load balancing some asterisk boxes for 
transcoding and some other services like voicemail. Also i am using 
SIPp for load testing


Now the question is that, while everything is working great with NAT 
(even when i am using SIPp for testing with low load) when i am about 
to test it with more than 150 calls or something i am starting to get 
the following error:


/kamailio[34549]: ERROR: rtpproxy [rtpproxy.c:2735]: 
force_rtp_proxy(): incorrect port 0 in reply from rtp proxy/



A guess, but each call takes 4 file descriptors, so you might be running 
into the normal os limit of 1024 open files per process.
Recent rtpproxy versions have the -L option to increase the limit. It 
should be 4 times the number of concurrent calls you expect to handle.


Again, just a guess.



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Re: [SR-Users] rtpproxy question

[Maxim Sobolev]

We don't support SRTP de/re-encryption at this point, neither in master nor
in 2.0. The work to add it is underway, but we are not quite there yet.
Pass-through mode should be working fine though, we've tested it recently
specifically.
On Jun 7, 2016 12:27 PM, "Albert Petit"  wrote:

> Hi ,
>
> Sorry for previous question. Finally it seems i got confused because
> traffic my UA was sending was not properly encrypted .Then when doing
> server tcpdump wireshark was seeing it still as RTCP (and not SRTCP) when
> jumping on rtpproxy :-) When fixed problem in user agent all traffic is
> SRTCP and rtpproxy just bridges it to the destination.
>
> I was thinking of some interaction because new feature at
> https://github.com/sipwise/mediaproxy-ng ( (Bridging between RTP and SRTP
> user agents) but i understand this feature to decode SRTP can only be
> enabled when mediaproxy-ng is used (that feature is not supported by
> rtpproxy,  but by mediaproxy-ng correct?).
>
>  BTW can this feature be enabled  easily when mediaproxy-ng is used? Is
> that enabled using following flags on rtpproxy_offer etc? (*s, S, p, P* -
> These flags control the RTP transport protocol that should be used towards
> the recipient of the SDP. If none of them are specified, the protocol given
> in the SDP is left untouched. Otherwise, the "S" flag indicates that SRTP
> should be used, while "s" indicates that SRTP should not be used. "P"
> indicates that the advanced RTCP profile with feedback messages should be
> used, and "p" indicates that the regular RTCP profile should be used. As
> such, the combinations "sp", "sP", "Sp" and "SP" select between RTP/AVP,
> RTP/AVPF, RTP/SAVP and RTP/SAVPF, respectively.)
>
> Thanks
> Albert
>
>
>
> 2016-06-01 12:18 GMT+02:00 Albert Petit :
>
>> Hi list
>>
>> I have installed a newer version of rtpproxy (2.0) in our development
>> server , i was happy because it brings lot of performance improvements
>>
>> However i have found an issue when clients use TLS+SRTP
>>
>> In that scenario i do not want rtpproxy to decrypt/reencrypt the traffic
>> as my B2BUA does that or in some scenarios we do plan to add end to end
>> encryption. RTPProxy 1.X was doing that perfectly
>>
>> Instead since i use rtpproxy 2.0 all SRTP and SRTCP traffic is decrypted
>> by rtpproxy and i receive it clear in the B2BUA
>>
>> How can i disable this new feature and that rtpproxy just keeps sending
>> the received SRTP/SRTCP stream without decrypting it?
>>
>>
>>
>>
>>
>>
>
>
> --
> Albert Petit
> Agile Software Architect 
> GENAKER - Esi Mobile Solutions SL
> www.genaker.net
> Phone +34 932 422 885
>
>
>
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Re: [SR-Users] rtpproxy question

[Albert Petit]

Hi ,

Sorry for previous question. Finally it seems i got confused because
traffic my UA was sending was not properly encrypted .Then when doing
server tcpdump wireshark was seeing it still as RTCP (and not SRTCP) when
jumping on rtpproxy :-) When fixed problem in user agent all traffic is
SRTCP and rtpproxy just bridges it to the destination.

I was thinking of some interaction because new feature at
https://github.com/sipwise/mediaproxy-ng ( (Bridging between RTP and SRTP
user agents) but i understand this feature to decode SRTP can only be
enabled when mediaproxy-ng is used (that feature is not supported by
rtpproxy,  but by mediaproxy-ng correct?).

 BTW can this feature be enabled  easily when mediaproxy-ng is used? Is
that enabled using following flags on rtpproxy_offer etc? (*s, S, p, P* -
These flags control the RTP transport protocol that should be used towards
the recipient of the SDP. If none of them are specified, the protocol given
in the SDP is left untouched. Otherwise, the "S" flag indicates that SRTP
should be used, while "s" indicates that SRTP should not be used. "P"
indicates that the advanced RTCP profile with feedback messages should be
used, and "p" indicates that the regular RTCP profile should be used. As
such, the combinations "sp", "sP", "Sp" and "SP" select between RTP/AVP,
RTP/AVPF, RTP/SAVP and RTP/SAVPF, respectively.)

Thanks
Albert



2016-06-01 12:18 GMT+02:00 Albert Petit :

> Hi list
>
> I have installed a newer version of rtpproxy (2.0) in our development
> server , i was happy because it brings lot of performance improvements
>
> However i have found an issue when clients use TLS+SRTP
>
> In that scenario i do not want rtpproxy to decrypt/reencrypt the traffic
> as my B2BUA does that or in some scenarios we do plan to add end to end
> encryption. RTPProxy 1.X was doing that perfectly
>
> Instead since i use rtpproxy 2.0 all SRTP and SRTCP traffic is decrypted
> by rtpproxy and i receive it clear in the B2BUA
>
> How can i disable this new feature and that rtpproxy just keeps sending
> the received SRTP/SRTCP stream without decrypting it?
>
>
>
>
>
>


-- 
Albert Petit
Agile Software Architect 
GENAKER - Esi Mobile Solutions SL
www.genaker.net
Phone +34 932 422 885
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Re: [SR-Users] rtpproxy question

[Daniel-Constantin Mierla]

Hello,

are you sure rtpproxy 2.0 does encryption/decryption of the RTP/SRTP? I
haven't noticed that the v2.0 has support for such feature.

What are the parameters you are using to control rtpproxy from kamailio.cfg?

Cheers,
Daniel


On 01/06/16 12:18, Albert Petit wrote:
> Hi list
>
> I have installed a newer version of rtpproxy (2.0) in our development
> server , i was happy because it brings lot of performance improvements 
>
> However i have found an issue when clients use TLS+SRTP
>
> In that scenario i do not want rtpproxy to decrypt/reencrypt the
> traffic as my B2BUA does that or in some scenarios we do plan to add
> end to end encryption. RTPProxy 1.X was doing that perfectly
>
> Instead since i use rtpproxy 2.0 all SRTP and SRTCP traffic is
> decrypted by rtpproxy and i receive it clear in the B2BUA
>
> How can i disable this new feature and that rtpproxy just keeps
> sending the received SRTP/SRTCP stream without decrypting it?
>
>
>
>
>
>
>
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-- 
Daniel-Constantin Mierla
http://www.asipto.com - http://www.kamailio.org
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] rtpproxy module question

[Daniel-Constantin Mierla]

Hello,

I am not familiar with the insights of rtpproxy source code, so I don't
know if there is a limitation for duration and for how long it is.

Cheers,
Daniel


On 27/05/16 10:42, gmele wrote:
>
> Hello,
>
>  
>
> Thx for the explanation, so it means that as soon as the callee
> connects to the RTP Proxy, the rtp proxy will use the callee ip
> address and port to forward the rtp stream and ignore the initial
> learned ip/port? Is there a duration limitation in this learning mode?
> Meaning that if the callee waits to much to send the first udp packet,
> the rtp proxy will use the ip/port set during negotiation?
>
>  
>
> Thx
>
>  
>
> Giovanni
>
>  
>
> *From:*Daniel-Constantin Mierla-6 [via SIP Router]
> [mailto:ml-node+[hidden email]
> ]
> *Sent:* vendredi 27 mai 2016 10:09
> *To:* Mele Giovanni
> *Subject:* Re: rtpproxy module question
>
>  
>
> Hello,
>
> initially the rtpproxy is in so called learning mode, waiting for the
> first rtp packet to come from each side of the call. Before receiving
> first rtp packet it relies on source ip of signaling.
>
> If the SDP has the device IP (you can eventually set that in the
> proxy), then you can use 'r' flag for rtpproxy_manage() to tell
> rtpproxy that it should trust the IP from sdp.
>
> Cheers,
> Daniel
>
>  
>
> On 27/05/16 09:55, gmele wrote:
>
> Hello,
>
>  
>
> we are using an RTP Proxy from rtpproxy.org as media relay to establish
>
> communication between our mobile phones. Of course, we are using the
>
> kamailio rtpproxy module to modify the SDP payload and control the proxy.
>
>  
>
> In our Kamailio configuration, we have 1 kamailio configured as Proxy and
>
> one kamailio configured as Registrar. So calls go through the Proxy and 
> then
>
> to the Registrar who will update the SDP header and select an available 
> rtp
>
> proxy.
>
>  
>
> We have noticed that sometimes, the rtp udp flow between the phones isn't
>
> routed properly by the rtpproxies, ending in the communication drop (all 
> the
>
> SIP nego is working well, and the SDP are correctly patched with the rtp
>
> proxy address and port). 
>
>  
>
> Analyzing the RTP proxy packets, we have found that the Kamailio registrar
>
> gives the Kamailio proxy ip address in the RTP proxy create session 
> command,
>
> but keeps the original sdp port.
>
> command looks like this:
>
> Uc96,101 DC -PbO~Wnm  
>
> PZU5OITCW;1 
>
>  
>
> We are using rtpproxy_manage() without any flags.
>
> It seems to us that this ip and port are used as default forward route as
>
> long as the callee hasnt connected to the rtpproxy. *Is it correct?
>
> *
>
> If its true, Its seems to us that this cant work as we are mixing the 
> proxy
>
> sip address with a udp port open on the phone ? *Is our analysis correct 
> ?*
>
>  
>
> Can we use some option in rtpproxy_manage to replace the proxy ip by the
>
> phone ip as seen in the via route ? 
>
>  
>
>  
>
> Thx for your help
>
>  
>
> Giovanni
>
>  
>
>  
>
>  
>
>  
>
>  
>
>  
>
>  
>
>  
>
>  
>
> --
>
> View this message in context: 
> http://sip-router.1086192.n5.nabble.com/rtpproxy-module-question-tp148850.html
>
> Sent from the Users mailing list archive at Nabble.com.
>
>  
>
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>
>
>
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>
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Re: [SR-Users] rtpproxy module question

[gmele]

Hello,

Thx for the explanation, so it means that as soon as the callee connects to the 
RTP Proxy, the rtp proxy will use the callee ip address and port to forward the 
rtp stream and ignore the initial learned ip/port? Is there a duration 
limitation in this learning mode? Meaning that if the callee waits to much to 
send the first udp packet, the rtp proxy will use the ip/port set during 
negotiation?

Thx

Giovanni

From: Daniel-Constantin Mierla-6 [via SIP Router] 
[mailto:ml-node+s1086192n148852...@n5.nabble.com]
Sent: vendredi 27 mai 2016 10:09
To: Mele Giovanni
Subject: Re: rtpproxy module question


Hello,

initially the rtpproxy is in so called learning mode, waiting for the first rtp 
packet to come from each side of the call. Before receiving first rtp packet it 
relies on source ip of signaling.

If the SDP has the device IP (you can eventually set that in the proxy), then 
you can use 'r' flag for rtpproxy_manage() to tell rtpproxy that it should 
trust the IP from sdp.

Cheers,
Daniel

On 27/05/16 09:55, gmele wrote:

Hello,



we are using an RTP Proxy from rtpproxy.org as media relay to establish

communication between our mobile phones. Of course, we are using the

kamailio rtpproxy module to modify the SDP payload and control the proxy.



In our Kamailio configuration, we have 1 kamailio configured as Proxy and

one kamailio configured as Registrar. So calls go through the Proxy and then

to the Registrar who will update the SDP header and select an available rtp

proxy.



We have noticed that sometimes, the rtp udp flow between the phones isn't

routed properly by the rtpproxies, ending in the communication drop (all the

SIP nego is working well, and the SDP are correctly patched with the rtp

proxy address and port).



Analyzing the RTP proxy packets, we have found that the Kamailio registrar

gives the Kamailio proxy ip address in the RTP proxy create session command,

but keeps the original sdp port.

command looks like this:

Uc96,101 DC -PbO~Wnm  

PZU5OITCW;1



We are using rtpproxy_manage() without any flags.

It seems to us that this ip and port are used as default forward route as

long as the callee hasnt connected to the rtpproxy. *Is it correct?

*

If its true, Its seems to us that this cant work as we are mixing the proxy

sip address with a udp port open on the phone ? *Is our analysis correct ?*



Can we use some option in rtpproxy_manage to replace the proxy ip by the

phone ip as seen in the via route ?





Thx for your help



Giovanni



















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Re: [SR-Users] rtpproxy module question

[Daniel-Constantin Mierla]

Hello,

initially the rtpproxy is in so called learning mode, waiting for the
first rtp packet to come from each side of the call. Before receiving
first rtp packet it relies on source ip of signaling.

If the SDP has the device IP (you can eventually set that in the proxy),
then you can use 'r' flag for rtpproxy_manage() to tell rtpproxy that it
should trust the IP from sdp.

Cheers,
Daniel


On 27/05/16 09:55, gmele wrote:
> Hello,
>
> we are using an RTP Proxy from rtpproxy.org as media relay to establish
> communication between our mobile phones. Of course, we are using the
> kamailio rtpproxy module to modify the SDP payload and control the proxy.
>
> In our Kamailio configuration, we have 1 kamailio configured as Proxy and
> one kamailio configured as Registrar. So calls go through the Proxy and then
> to the Registrar who will update the SDP header and select an available rtp
> proxy.
>
> We have noticed that sometimes, the rtp udp flow between the phones isn't
> routed properly by the rtpproxies, ending in the communication drop (all the
> SIP nego is working well, and the SDP are correctly patched with the rtp
> proxy address and port). 
>
> Analyzing the RTP proxy packets, we have found that the Kamailio registrar
> gives the Kamailio proxy ip address in the RTP proxy create session command,
> but keeps the original sdp port.
> command looks like this:
> Uc96,101 DC -PbO~Wnm  
> PZU5OITCW;1 
>
> We are using rtpproxy_manage() without any flags.
> It seems to us that this ip and port are used as default forward route as
> long as the callee hasnt connected to the rtpproxy. *Is it correct?
> *
> If its true, Its seems to us that this cant work as we are mixing the proxy
> sip address with a udp port open on the phone ? *Is our analysis correct ?*
>
> Can we use some option in rtpproxy_manage to replace the proxy ip by the
> phone ip as seen in the via route ? 
>
>
> Thx for your help
>
> Giovanni
>
>
>
>
>
>
>
>
>
> --
> View this message in context: 
> http://sip-router.1086192.n5.nabble.com/rtpproxy-module-question-tp148850.html
> Sent from the Users mailing list archive at Nabble.com.
>
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Re: [SR-Users] RTPPROXY error message

[Daniel-Constantin Mierla]

Hello,

if rtpproxy is listening on udp:127.0.0.1:7722, then you may have some
limits in place regarding sockets/traffic -- if you have selinux
enabled, try without it. Also, you can try by lowering the value for
children parameter.

Regarding the SQL errors, it seems you run the config from Siremis, but
you haven't created the stored procedures required there.

Cheers,
Daniel

On 14/03/16 23:17, Kevin Greene wrote:
> Hi:
>
> I am sure this has been answered before on this list and I have tried the 
> basic
> answers I got from Google without any luck. Any suggests would be great.
>
> The Kamailio host is running Oracle Linux 6.7 (64 bit) and is fully patched as
> of today.
> The Kamailio version is:
>
> Server:: kamailio (4.3.4 (x86_64/linux))
> Build:: mi_core.c compiled on 14:25:16 Nov 25 2015 with gcc 4.4.7
>
> I have two major issues:
>
> I am seeing this on my Kamailio log:
>
> Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: ERROR: rtpproxy
> [rtpproxy.c:1681]: send_rtpp_command(): can't send command to a RTP proxy
> Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: ERROR: rtpproxy
> [rtpproxy.c:1716]: send_rtpp_command(): proxy  does not
> respond, disable it
> Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: WARNING: rtpproxy
> [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP proxy
> Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: WARNING: rtpproxy
> [rtpproxy.c:1610]: rtpp_test(): support for RTP proxy  has
> been disabled temporarily
>
> So the initial thought is that the proxy is not running, however I can see the
> proxy is alive:
>
> [root@sipserver boot]# ps -All | grep rtpproxy
> 5 S   498  2843 1  1  80   0 - 32524 hrtime ?01:29:06 rtpproxy
>
> When I do "netstat -a", I can see 24 established connections.
> This is a small sample:
>
> udp0  0 localhost:47937 localhost:rtpproxy 
> ESTABLISHED
> udp0  0 localhost:38746 localhost:rtpproxy 
> ESTABLISHED
> udp0  0 localhost:33660 localhost:rtpproxy 
> ESTABLISHED
>
> So I am not sure what is going on.
>
> The second problem:
>
> I see this in the Kamailio log:
>
> Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: db_mysql
> [km_dbase.c:124]: db_mysql_submit_query(): driver error on query: PROCEDURE
> kamailio.kamailio_rating does not exist (1305)
> Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: 
> [db_query.c:181]: db_do_raw_query(): error while submitting query
> Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: sqlops
> [sql_api.c:265]: sql_do_query(): cannot do the query [call
> kamailio_rating('default')]
>
> When I tried to create the database originally, the script that was part of 
> the
> package would not work. I therefore manually created the database so I am
> thinking that something got left out. I have been over the schema and I am not
> sure what is missing. The database is MySQL (5.5) running in a separate host.
>
> Does anyone have an idea where to look what is missing? I was told that there 
> is
> some sort of script that can "repair" the Kamailio schema in the database but 
> I
> have not been able to find this magical script.
>
> Thanks.
>
> Kevin
>
>
>
>
>
>
>
>
>
>
>
>
>
>
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Re: [SR-Users] RTPPROXY issue and sip to sip calling

[SamyGo]

Hi Rehan,

No matter which mode you are running rtpproxy in that IP will always be the
IP of the machine it is running on.
That means that SDP will take that IP once routed to locally subnet A2B
servers.
As far as the A2B detecting SIP user as online or offline based on DB,  I
am not too sure about it. If it is realtime then I think it should work out
of box. You may need to try it out to know it accurately.

Regards,
Sammy
On Jan 30, 2016 02:05, "Ahmed Rehan"  wrote:

> Dear All
>
> I m trying to setup kamailio and asterisk in load balancing with a2billing
> . Currently all of my VMs, one Kamailio and two asterisks are on same
> subnet . I have started the RTPproxy like below
>
> ./rtpproxy -s udp:127.0.0.1:7722 -l X.X.X.153 -m 1 -M 5 -u root
> root -F -d INFO LOG_LOCAL0
>
> My question is if all the VMs are on same subnet with same gateway what
> should be written in the private IP X.X.X.153/
>
> Secondly i m authenticating and registering the SIP on kamailio using the
> A2B DB . all the dialplan for a2b is being run on asterisk . Now if i want
> to call SIP peer to Peer like in case of followme case ,
>
> How should i route the calls in Kamailio ? will it be using usr loc
> module? if so any help will be appreciated
>
> --
>
>
> Regards
> Ahmed Rehan
>
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Re: [SR-Users] rtpproxy does not response, disable it

[smititelu]

I solved this problem by changing the 127.0.0.1 in rtpengine startup 
config into 10.109.247.90 which is the eth0 address of the rtpengine 
machine.


Now Kamailio successfully connect to the rtpengine daemon.


Nice you solved it. Didn't think of it :D
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Re: [SR-Users] rtpproxy does not response, disable it

[smititelu]

I’ve already set modparam("rtpengine", "force_send_interface", 
“10.109.247.80”) in my kamailio.cfg from the very beginning.
Can you try force ping from kamailio machine (.80) to rtpengine machine 
(.90), on 10.109.247.80 interface? (it should work)



There is nothing prompt out on command “kamctl fifo nh_show_rtpp all”
If there is nothing shown it means that kamailio sees no rtpengine 
nodes. What kamailio/rtpengine version are you using? (are they both 
latest upstream?!). The sock line should be fine: modparam("rtpengine", 
"rtpengine_sock", "udp:10.109.247.90:7723")


Can you increase the log level(debug=3) and send some debug info on 
what's happening right after you restart kamailio?(for rtpengine module)


There is nothing captured on "ngerp -W byline -d em1 port 5095” on 
Kamilio machine.
Wehn eventually you see something for the show ctl command, you should 
try capturing for dst port 7723, on kamailio machine, to see the 
commands sent to rtpengine.


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Re: [SR-Users] rtpproxy does not response, disable it

[Zodiac]

I solved this problem by changing the 127.0.0.1 in rtpengine startup config 
into 10.109.247.90 which is the eth0 address of the rtpengine machine.

Now Kamailio successfully connect to the rtpengine daemon.



> 在 2015年12月22日，17:38，Zodiac  写道：
> 
> Hi, I am very glad that you can answer me for that.
> 
> I’ve already set modparam("rtpengine", "force_send_interface", 
> “10.109.247.80”) in my kamailio.cfg from the very beginning.
> 
> The port 7723 on 10.109.247.90 which rtpengine daemon runs is not blocked by 
> firewall.
> 
> There is nothing prompt out on command “kamctl fifo nh_show_rtpp all”
> 
> There is nothing captured on "ngerp -W byline -d em1 port 5095” on Kamilio 
> machine.
> 
> There is an prompt “404 rtpproxy not found” on command “kamctl fifo 
> nh_enable_rtpp ump:10.109.247.90:7723 1”
> 
> What other reasons can there be for my case?
> 
> Furthermore, 10.109.247.80(Kamailio) and 10.109.247.90(rtpengine daemon) are 
> both private IP address, not public address. Dose this affect?
> 
> 
> This is the kamailio.cfg portion:
> 
> 
> 
> This is the rtpengine daemon startup config:
> 
> 
> 
>> 在 2015年12月21日，23:53，smititelu > > 写道：
>> 
>> Hi Zodiac,
>> 
>> 1. Can you see your configured rtp node on: "kamctl fifo nh_show_rtpp all" ?
>> 2. Can you ngrep the commands being sent by kamailio to rtpengine?(on both 
>> kamailio and rtpengine machine)
>> 3. Do you have some firewalling rules that may block that 7723 port?
>> 
>> Stefan
>> 
>> On 21.12.2015 17:45, Zodiac wrote:
>>> Dear friends,
>>> I am working on a program on Kamailio and rtpengine proxy. I am wondering 
>>> whether can I set Kamailio and rtpengine daemon on different physical 
>>> machines. For example, I set Kamailio on a machine with IP 
>>> address:10.109.247.80, and launch rtpengine daemon on another machine with 
>>> interface parameter as 10.109.247.90 and ng port 7723. I set parameter in 
>>> Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, 
>>> “udp:10.109.247.90:7723”).
>>> 
>>> Unfortunately I got debug message like this:
>>> 
>>> ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send 
>>> command to a RTP proxy
>>> 
>>> ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy 
>>>  does not respond, disable it
>>> 
>>> ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to 
>>> ping
>>> 
>>> And, I also tried to set Kamailio and rtpengine daemon in a same 
>>> machine,and use modparam(“rtpengine”, “rtpengine_sock”, 
>>> “udp:localhost:7723”). And Kamailio can work functionally under this 
>>> situation. rtpengine daemon can receive ping message from Kamailio and 
>>> rtpengine daemon can work as suspected. So for the later case, is it 
>>> supposed that Kamailio be in the same machine with same localhost address? 
>>> Otherwise, what’s the reason for my ERROR?
>>> 
>>> 
>>> 
>>> 北京邮电大学网络技术研究院
>>> 网络与交换技术国家重点实验室
>>> 田军
>>> +86 18810315790
>>> mozillaf...@bupt.edu.cn 
>>> 
>>> 
>>> 
>>> 
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>> 
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> +86 18810315790
> mozillaf...@bupt.edu.cn 
> 
> 
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Re: [SR-Users] rtpproxy does not response, disable it

[Zodiac]

Hi, I am very glad that you can answer me for that.

I’ve already set modparam("rtpengine", "force_send_interface", “10.109.247.80”) 
in my kamailio.cfg from the very beginning.

The port 7723 on 10.109.247.90 which rtpengine daemon runs is not blocked by 
firewall.

There is nothing prompt out on command “kamctl fifo nh_show_rtpp all”

There is nothing captured on "ngerp -W byline -d em1 port 5095” on Kamilio 
machine.

There is an prompt “404 rtpproxy not found” on command “kamctl fifo 
nh_enable_rtpp ump:10.109.247.90:7723 1”

What other reasons can there be for my case?

Furthermore, 10.109.247.80(Kamailio) and 10.109.247.90(rtpengine daemon) are 
both private IP address, not public address. Dose this affect?


This is the kamailio.cfg portion:



This is the rtpengine daemon startup config:



> 在 2015年12月21日，23:53，smititelu  写道：
> 
> Hi Zodiac,
> 
> 1. Can you see your configured rtp node on: "kamctl fifo nh_show_rtpp all" ?
> 2. Can you ngrep the commands being sent by kamailio to rtpengine?(on both 
> kamailio and rtpengine machine)
> 3. Do you have some firewalling rules that may block that 7723 port?
> 
> Stefan
> 
> On 21.12.2015 17:45, Zodiac wrote:
>> Dear friends,
>> I am working on a program on Kamailio and rtpengine proxy. I am wondering 
>> whether can I set Kamailio and rtpengine daemon on different physical 
>> machines. For example, I set Kamailio on a machine with IP 
>> address:10.109.247.80, and launch rtpengine daemon on another machine with 
>> interface parameter as 10.109.247.90 and ng port 7723. I set parameter in 
>> Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, 
>> “udp:10.109.247.90:7723”).
>> 
>> Unfortunately I got debug message like this:
>> 
>> ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command 
>> to a RTP proxy
>> 
>> ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy 
>>  does not respond, disable it
>> 
>> ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to 
>> ping
>> 
>> And, I also tried to set Kamailio and rtpengine daemon in a same machine,and 
>> use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And 
>> Kamailio can work functionally under this situation. rtpengine daemon can 
>> receive ping message from Kamailio and rtpengine daemon can work as 
>> suspected. So for the later case, is it supposed that Kamailio be in the 
>> same machine with same localhost address? Otherwise, what’s the reason for 
>> my ERROR?
>> 
>> 
>> 
>> 北京邮电大学网络技术研究院
>> 网络与交换技术国家重点实验室
>> 田军
>> +86 18810315790
>> mozillaf...@bupt.edu.cn 
>> 
>> 
>> 
>> 
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Re: [SR-Users] rtpproxy does not response, disable it

[smititelu]

And, I also tried to set Kamailio and rtpengine daemon in a same 
machine,and use modparam(“rtpengine”, “rtpengine_sock”, 
“udp:localhost:7723”). And Kamailio can work functionally under this 
situation. rtpengine daemon can receive ping message from Kamailio and 
rtpengine daemon can work as suspected. So for the later case, is it 
supposed that Kamailio be in the same machine with same localhost 
address? Otherwise, what’s the reason for my ERROR?

You can try make use of the below parameter:

# Which local interface/IP should be used for the control channel
modparam("rtpengine", "force_send_interface", "X.X.X.X")

Thanks,
Stefan

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Re: [SR-Users] rtpproxy does not response, disable it

[smititelu]

Hi Zodiac,

1. Can you see your configured rtp node on: "kamctl fifo nh_show_rtpp all" ?
2. Can you ngrep the commands being sent by kamailio to rtpengine?(on 
both kamailio and rtpengine machine)

3. Do you have some firewalling rules that may block that 7723 port?

Stefan

On 21.12.2015 17:45, Zodiac wrote:

Dear friends,
I am working on a program on Kamailio and rtpengine proxy. I am 
wondering whether can I set Kamailio and rtpengine daemon on different 
physical machines. For example, I set Kamailio on a machine with IP 
address:10.109.247.80, and launch rtpengine daemon on another machine 
with interface parameter as 10.109.247.90 and ng port 7723. I set 
parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, 
“udp:10.109.247.90:7723”).


Unfortunately I got debug message like this:

ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send 
command to a RTP proxy


ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy 
 does not respond, disable it


ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not 
respond to ping


And, I also tried to set Kamailio and rtpengine daemon in a same 
machine,and use modparam(“rtpengine”, “rtpengine_sock”, 
“udp:localhost:7723”). And Kamailio can work functionally under this 
situation. rtpengine daemon can receive ping message from Kamailio and 
rtpengine daemon can work as suspected. So for the later case, is it 
supposed that Kamailio be in the same machine with same localhost 
address? Otherwise, what’s the reason for my ERROR?




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Re: [SR-Users] rtpproxy does not response, disable it

[Alex Balashov]

Yes, you can certainly run rtpengine on a different host than the 
Kamailio host.


Is it possible there are firewall rules on the remote rtpengine host 
that are blocking UDP connections from your Kamailio box to port 7723? 
That would be my first thought.


-- Alex


--
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303 Perimeter Center North, Suite 300
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Re: [SR-Users] RTPPROXY unpredictable behavior

[Carsten Bock]

Hi Javohir,

don't worry, your english is fine.
Can you send us the parameters you have set for rtpproxy in your kamailio cfg?
You should find something like this:

modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:2")

Make sure it matches the configuration of the RTPProxy.

Thanks,
Carsten

2015-12-09 11:32 GMT+01:00 Javohir Maxmarajabov :
> I am trying to enable RTPPROXY on debian, but it seems RTPProxy is ignoring
> my arguments...
>
> So, I am launching RTPPROXY using command:
>
> rtpproxy -l _MY_PUBLIC_IP_ -s udp:127.0.0.1 7722 -p /var/run/rtpproxy.pid -R
> -a -P -r /tmp/rtppath -S /tmp/rtpspool -u rtpproxy rtpproxy
>
> Also tried with:
>
> rtpproxy -l _MY_PUBLIC_IP_ -s udp:127.0.0.1 7722 -u rtpproxy rtpproxy
>
> And when I try to start Kamailio service I am getting error:
>
> 0(10562) ERROR: rtpproxy [rtpproxy.c:1681]: send_rtpp_command(): can't send
> command to a RTP proxy
>
> 0(10562) ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy does
> not respond, disable it
>
> 0(10562) WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version
> of the RTP proxy
>
> 0(10562) WARNING: rtpproxy [rtpproxy.c:1610]: rtpp_test(): support for RTP
> proxy has been disabled temporarily
>
> p.s. I have checked RTPPROXY, and I am sure that it is launched
>
> ps aux | grep rtpproxy
>
> returned me:
>
> root 2770 0.0 0.0 19392 480 ? Ssl 04:59 0:00 /usr/bin/rtpproxy -l
> _MY_PUBLIC_IP_ -s udp:127.0.0.1 7722 -p /var/run/rtpproxy.pid -R -a -P -r
> /tmp/rtppath -S /tmp/rtpspool -u rtpproxy rtpproxy
>
> root 2775 0.0 0.0 11740 932 pts/1 S+ 04:59 0:00 grep --color=auto rtp
>
> Which says everything is ok and rtpproxy is running with arguments that I
> declared
>
> But the problem is, When I check it with:
>
> netstat -tulpn | grep rtpproxy
>
> I am receiving:
>
> udp 0 0 127.0.0.1:2 0.0.0.0:* 2853/rtpproxy
>
> As you can see rtpproxy is listening on port 2 and completely ignoring
> arguments I declared :(
>
> Where and What am I doing wrong?
>
> Sorry for my English and Thanks in advance...
>
>
> --
> Javokhir M.M.
> 
>
> Disclaimer: The information in this e-mail is confidential. If you are not
> addressed recipient then please return and delete this e-mail from your
> system. Unauthorised use of or disclose the contents of this e-mail may be
> unlawful.
>
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Re: [SR-Users] rtpproxy database structure for kamailio 4.2.3

[Jan Hazenberg]

Daniel,

Clear, i will simply add the field until we can upgrade. Thanks for the 
clarification.


Jan

Daniel-Constantin Mierla schreef op 2015-11-19 13:38:

Hello,

it was a different value for column name in the code than in the 
default

rtpproxy table. The definition of the table used setid and the code
set_name. I pushed a fix (in master, 4,3 and 4,2 branches), so the code
is coherent with the database table.

If you don't want to upgrade to latest version in the branch, then have
the set_name column with the same definition of setid column.

Although the type of the column is varchar, it expect to be a number --
from the commit I assumed the developer wanted to extend to non-numeric
set ids, but hasn't done it so far.

If you are not using set ids in the config, you can set the value for
setid/set_name columns to "0".

You can select specific rtpproxy by grouping with a set id -- see:

http://kamailio.org/docs/modules/4.3.x/modules/rtpproxy.html#idp15531672

Cheers,
Daniel

On 19/11/15 12:54, Jan Hazenberg wrote:

Hi all,

I'm trying to load the rtpproxy url's from database but get a error
when starting kamailio (4.2.3):

ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver
error on query: Unknown column 'set_name' in 'field list'

Looks like the field set_name is missing in the database structure,
but i can't find it in the docs. If i add the field kamailio starts,
but what is the field set_name used for. Seems it needs to be a int
value:

ERROR: rtpproxy [rtpproxy.c:528]: get_rtpp_set(): Invalid set name
'test' - must be integer


Thanks,

Jan




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Re: [SR-Users] rtpproxy database structure for kamailio 4.2.3

[Daniel-Constantin Mierla]

Hello,

it was a different value for column name in the code than in the default
rtpproxy table. The definition of the table used setid and the code
set_name. I pushed a fix (in master, 4,3 and 4,2 branches), so the code
is coherent with the database table.

If you don't want to upgrade to latest version in the branch, then have
the set_name column with the same definition of setid column.

Although the type of the column is varchar, it expect to be a number --
from the commit I assumed the developer wanted to extend to non-numeric
set ids, but hasn't done it so far.

If you are not using set ids in the config, you can set the value for
setid/set_name columns to "0".

You can select specific rtpproxy by grouping with a set id -- see:

http://kamailio.org/docs/modules/4.3.x/modules/rtpproxy.html#idp15531672

Cheers,
Daniel

On 19/11/15 12:54, Jan Hazenberg wrote:
> Hi all,
>
> I'm trying to load the rtpproxy url's from database but get a error
> when starting kamailio (4.2.3):
>
> ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver
> error on query: Unknown column 'set_name' in 'field list'
>
> Looks like the field set_name is missing in the database structure,
> but i can't find it in the docs. If i add the field kamailio starts,
> but what is the field set_name used for. Seems it needs to be a int
> value:
>
> ERROR: rtpproxy [rtpproxy.c:528]: get_rtpp_set(): Invalid set name
> 'test' - must be integer
>
>
> Thanks,
>
> Jan
>
>
>
>
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Re: [SR-Users] RtpProxy socket timeout not working

[Daniel-Constantin Mierla]

Hello,

the code shows that is added for U command, if to-tag exists:

https://github.com/kamailio/kamailio/blob/master/modules/rtpproxy/rtpproxy.c#L2692

Can you edit the rtpproxy.c and add a log message (e.g., using
LM_ERR("..."); ) there and is it is printed by kamailio?

Cheers,
Daniel

On 17/10/15 20:57, Arun Kumar wrote:
> Hi
>
> Rtp Session timeout notification is not working , we tried with
> kamailio 4.2 with rtpproxy 2.0 / 1.3 beta/1.2 releases , i think
> kamailio rtpproxy is not submitting session timeout socket to rtpproxy 
>
>
>  kamailio 4.2.0
> ---
>  
> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722
> ")
> modparam("rtpproxy", "timeout_socket", "xmlrpc:http://127.0.0.1:5060/RPC";)
>
> ...
> rtpproxy_manage();
> 
>
>
> RtpProxy
> -
> RTPProxy 2.0
> RTPProxy 1.3 Beta with patch given by /Daniel/
> RTPProxy 1.2 
>
> rtpproxy   -F   -s udp:127.0.0.1:7722  -d DBUG
> -f -i -T 10
>
>
> RTPRoxy Log
> 
> DBUG:get_command: received command "4242_9 Uc0,8,101 2479e562440c023d
> 192.168.1.103 6074 5410ef73;1"
> INFO:handle_command: new session 2479e562440c023d, tag 5410ef73;1
> requested, type strong
> INFO:handle_command: new session on a port 45984 created, tag 5410ef73;1
> INFO:handle_command: pre-filling caller's address with
> 192.168.1.103:6074 
> DBUG:doreply: sending reply "4242_9 45984
> "
> DBUG:get_command: received command "4242_10 Lc0,101 2479e562440c023d
> 192.168.1.102 4002 5410ef73;1 ADqH-A-mLLN0th2gGoopYMK1cy.-ozDf;1"
> INFO:handle_command: lookup on ports 45984/36756, session timer restarted
> INFO:handle_command: pre-filling callee's address with
> 192.168.1.102:4002 
> DBUG:doreply: sending reply "4242_10 36756
> "
> INFO:rxmit_packets: callee's address latched in: 192.168.1.102:4002
>  (RTP)
> INFO:rxmit_packets: callee's address latched in: 192.168.1.102:4003
>  (RTCP)
> INFO:rxmit_packets: caller's address latched in: 192.168.1.103:6075
>  (RTCP)
> INFO:rxmit_packets: caller's address latched in: 192.168.1.103:6074
>  (RTP)
> INFO:process_rtp: session timeout
> INFO:remove_session: RTP stats: 328 in from callee, 5 in from caller,
> 333 relayed, 0 dropped
> INFO:remove_session: RTCP stats: 3 in from callee, 4 in from caller, 7
> relayed, 0 dropped
> INFO:remove_session: session on ports 45984/36756 is cleaned up
>
>
> Kamailio RTP log
> ---
>  3(4242) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type():
> type  found valid
>  3(4242) DEBUG: rtpproxy [rtpproxy.c:2704]: force_rtp_proxy(): proxy
> reply: 45984
>
> for more log please see attachment 
>
>
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Re: [SR-Users] RTPProxy incorrect port 0 in reply from rtp proxy

[Grant Bagdasarian]

Looks like I forgot to call rtpproxy_manage() for BYE and CANCEL requests.
netstat -unlp showed a lot of port being used by rtpproxy even after the load 
test finished.

The ports weren't being freed and that's probably the reason it gave the error. 
It was running out of ports to use.

From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Grant Bagdasarian
Sent: Wednesday, August 12, 2015 10:55 AM
To: Kamailio (SER) - Users Mailing List (sr-users@lists.sip-router.org) 

Subject: [SR-Users] RTPProxy incorrect port 0 in reply from rtp proxy

Hello,

I've been load testing kamailio(4.2) with rtpproxy(2.0) using sipp, and when I 
hit 50+ CPS I start getting the following error: "incorrect port 0 in reply 
from rtp proxy".
I'm running 4 instances of rtpproxy on the same instance as kamailio.

Could somebody tell me why I'm receiving this error? Do I need to run more 
instances of rtpproxy?

# <#--RTPPROXY Parameters--#>
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7721 udp:127.0.0.1:7722 
udp:127.0.0.1:7723 udp:127.0.0.1:7724")

In request_route:
if (is_method("INVITE")) {
   record_route();

   if (has_body("application/sdp")) {
   rtpproxy_manage();
   }
}

In onreply_route:

if (has_body("application/sdp")) {
   rtpproxy_manage();
}


rtpproxy -s udp:127.0.0.1:7721 -u rtpproxy rtpproxy -p 
/var/run/rtpproxy/rtpproxy.pid -l 10.0.0.1
rtpproxy -s udp:127.0.0.1:7722 -u rtpproxy rtpproxy -p 
/var/run/rtpproxy/rtpproxy1.pid -l 10.0.0.1
rtpproxy -s udp:127.0.0.1:7723 -u rtpproxy rtpproxy -p 
/var/run/rtpproxy/rtpproxy2.pid -l 10.0.0.1
rtpproxy -s udp:127.0.0.1:7724 -u rtpproxy rtpproxy -p 
/var/run/rtpproxy/rtpproxy3.pid -l 10.0.0.1

Regards,

Grant
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Re: [SR-Users] rtpproxy 2.0

[Fred Posner]

Have tested it on virtual and physical. Works well, no need to patch for VM or 
advertised address. Had no complaints from users with 1.2 and none since 2.0 

Installed from git for testing. 

-- Fred

> On May 22, 2015, at 4:37 AM, Klaus Darilion  
> wrote:
> 
> Hi!
> 
> I just found out that there is a new rtpproxy release:
> http://www.rtpproxy.org/post/v2release/
> 
> Has anybody tested it and want to share some experiences? Or have people
> turned to rtpengine meanwhile? (I am still using rtpproxy 1.2 as I do
> not need new features).
> 
> regards
> Klaus
> 
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Re: [SR-Users] rtpproxy 2.0

[Øyvind Kolbu]

On 22.05.2015 10:37, Klaus Darilion wrote:

I just found out that there is a new rtpproxy release:
http://www.rtpproxy.org/post/v2release/

Has anybody tested it and want to share some experiences? Or have people
turned to rtpengine meanwhile? (I am still using rtpproxy 1.2 as I do
not need new features).



Upgraded a few servers from RHEL6 to RHEL7, in order to try rtpengine, 
and then got
the chance to briefly test rtpproxy 2.0 as it is default in RHEL7. It 
just works fine, haven't
tested any new options. Only change/improvement we've found is that on 
VMWare the

process is no longer running continuously at 100% CPU.

--
Øyvind Kolbu



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Re: [SR-Users] rtpproxy-ng and late SDP

[Richard Fuchs]

On 18/05/15 03:53 AM, Sebastian Damm wrote:

Hi Alex,

On Thu, May 14, 2015 at 5:47 AM, Alex Balashov
mailto:abalas...@evaristesys.com>> wrote:

According to the rtpengine module documentation for
rtpproxy_manage(), that's exactly what rtpproxy_manage() does:



http://kamailio.org/docs/modules/4.2.x/modules/rtpengine.html#rtpengine.f.rtpengine_manage

i.e.

- If INVITE with SDP, then do rtpengine_offer()
- If INVITE with SDP, when the tm module is loaded, mark transaction
with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for
rtpengine_answer()
- If ACK with SDP, then do rtpengine_answer()
- If reply with SDP to INVITE having code 1xx and 2xx, then do
rtpengine_answer() if the request had SDP or tm is not loaded,
otherwise do rtpengine_offer()



What I'm missing there is how to handle an INVITE without SDP. Does that
mean, that if we haven't called rtpproxy_manage() on the INVITE, the 1xx
or 2xx reply will trigger an rtpproxy_offer instead of rtpproxy_answer?


That's what's supposed to happen, yes.

Cheers

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Re: [SR-Users] rtpproxy-ng and late SDP

[Sebastian Damm]

Hi Alex,

On Thu, May 14, 2015 at 5:47 AM, Alex Balashov 
wrote:

> According to the rtpengine module documentation for rtpproxy_manage(),
> that's exactly what rtpproxy_manage() does:
>
>
>
> http://kamailio.org/docs/modules/4.2.x/modules/rtpengine.html#rtpengine.f.rtpengine_manage
>
> i.e.
>
> - If INVITE with SDP, then do rtpengine_offer()
> - If INVITE with SDP, when the tm module is loaded, mark transaction with
> internal flag FL_SDP_BODY to know that the 1xx and 2xx are for
> rtpengine_answer()
> - If ACK with SDP, then do rtpengine_answer()
> - If reply with SDP to INVITE having code 1xx and 2xx, then do
> rtpengine_answer() if the request had SDP or tm is not loaded, otherwise do
> rtpengine_offer()
>


What I'm missing there is how to handle an INVITE without SDP. Does that
mean, that if we haven't called rtpproxy_manage() on the INVITE, the 1xx or
2xx reply will trigger an rtpproxy_offer instead of rtpproxy_answer?

Best Regards,
Sebastian
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Re: [SR-Users] rtpproxy-ng and late SDP

[Alex Balashov]

Hello Sebastian,

On 05/13/2015 10:12 AM, Sebastian Damm wrote:


Is there a way to have rtpproxy_manage() handle those calls
automatically? I was thinking of setting a flag in the request which
is used by the rtpproxy_manage() function later.


According to the rtpengine module documentation for rtpproxy_manage(), 
that's exactly what rtpproxy_manage() does:



http://kamailio.org/docs/modules/4.2.x/modules/rtpengine.html#rtpengine.f.rtpengine_manage

i.e.

- If INVITE with SDP, then do rtpengine_offer()
- If INVITE with SDP, when the tm module is loaded, mark transaction 
with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for 
rtpengine_answer()

- If ACK with SDP, then do rtpengine_answer()
- If reply with SDP to INVITE having code 1xx and 2xx, then do 
rtpengine_answer() if the request had SDP or tm is not loaded, otherwise 
do rtpengine_offer()


-- Alex

--
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303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] [RTPproxy] Re: [OpenSIPS-Users] 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: [SR-Users] RTPProxy issue?

[Igor Potjevlesch]

Hello Maxim,

 

Just to let you know that I quick-fixed this issue by increasing the value 
compared into "ts_less" in rtp.c. Instead of the bitwise calculation, I set the 
maximum value of an unsigned integer.

 

I will schedule to test, qualify and replace all my instances with the new 
version. I will let you know if I reproduce the issue with the new version.

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de 
Maxim Sobolev
Envoyé : lundi 9 mars 2015 19:09
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Igor, yes, I'd say give 2.0 a try and see if the problem is still there. There 
were tons of changes, particularly in the rtp_resize subsystem.

Thanks!

 

On Sun, Mar 8, 2015 at 7:31 AM, Igor Potjevlesch mailto:igor.potjevle...@gmail.com> > wrote:

Hello Maxim,

 

I'm running legacy 1.2 or 1.4, not sure.

I see in the latest code that the function is still there. Do you suggest to 
upgrade or there's a patch to make?

 

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org 
<mailto:sr-users-boun...@lists.sip-router.org> ] De la part de Maxim Sobolev
Envoyé : samedi 7 mars 2015 09:14


À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Ah, ok, I see now. I did not realize you guys are using resizer. Which version 
of the software are you actually using? I.e. is it latest rel_2_0 / master, or 
some legacy 1.x code? We've done quite some revamping down there, so that it 
needs to be checked against the very latest code to make sure. Let us know.

Thanks!

On Mar 6, 2015 12:31 AM, "Igor Potjevlesch" mailto:igor.potjevle...@gmail.com> > wrote:

Hi Maxim,

 

Hard to do because it's in production.

I have a serious finding since yesterday on how this happened. 

 

My understanding is that the function "ts_less" returns FALSE into 
"rtp_resizer.c" because the timestamp between the two packets is > (1 << 31) 
[for example: 3740425320].

That's result in a drop of any following packets as I can see it into a capture.

 

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org 
<mailto:sr-users-boun...@lists.sip-router.org> ] De la part de Maxim Sobolev
Envoyé : vendredi 6 mars 2015 07:44
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Hi Igor, that's bit strange, since the rtpproxy is not checking any of the rtp 
flags including marker bit. It would help if you can post a tcpdump capture of 
the streams in question along with the log output of the rtpproxy running at 
the "dbug" level. Thanks!

On Mar 5, 2015 5:54 AM, "Igor Potjevlesch" mailto:igor.potjevle...@gmail.com> > wrote:

I reviewed again a call trace and I can be more precise: a RTP packet comes 
with a new SSRC and the Marker bit set to "True". This packet is properly 
forwarded.

 

Then, just after this packet, another RTP packet containing a new SSRC with the 
huge timestamp and the Marker bit set to "True" is coming from the UA. 

The RTPProxy stops forward since this packet.

 

Regards,

 

Igor.

 

De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com 
<mailto:igor.potjevle...@gmail.com> ] 
Envoyé : jeudi 5 mars 2015 11:34
À : mico...@gmail.com <mailto:mico...@gmail.com> ; 'Kamailio (SER) - Users 
Mailing List'
Objet : RE: [SR-Users] RTPProxy issue?

 

Hello,

 

Thank you.

 

Just to let you know, the RTPProxy is running in bridging mode.

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de 
Daniel-Constantin Mierla
Envoyé : jeudi 5 mars 2015 09:33
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Hello,

maybe Maxim (cc-ed) will be able to provide more insights.

Cheers,
DAniel

On 04/03/15 16:59, Igor Potjevlesch wrote:

Hello,

 

I discovered an issue related to the handling of "timestamp" and/or "Marker 
bit" with rtpproxy (I use the latest Extension 20081224).

 

The call-flow is the following: one UA places a call to A and put this call on 
hold. Then, the same UA call another number B. Individual streams are ok.

When the UA tries to transfer A with B, the RTPProxy receive a RTP packet with 
a huge timestamp and the Marker bit set to "True".

 

Just after this RTP packet, RTPProxy stop forward the RTP packets from A to B. 
B to C is still working.

 

Anyone have an idea?

Regards,

 

Igor.

 

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

[Zheng Frank]

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
>>>
>>>
>>
>> --
>> Sent from iPhone 6
>>
>> --
>> 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 post to this group, send email to rtppr...@googlegroups.com.
>> To view this discussion on the web visit
>> https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpUhu9fpwmqpRFiaXsQo9_%2B6M8MFtJ2sKi4kd5sr%3Ds%2BR5Q%40mail.gmail.com
>> 
>> .
>> For more options, visit https://groups.google.com/d/optout.
>>
>
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Re: [SR-Users] [RTPproxy] Re: [OpenSIPS-Users] 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
>>
>>
>
> --
> Sent from iPhone 6
>
> --
> 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 post to this group, send email to rtppr...@googlegroups.com.
> To view this discussion on the web visit
> https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpUhu9fpwmqpRFiaXsQo9_%2B6M8MFtJ2sKi4kd5sr%3Ds%2BR5Q%40mail.gmail.com
> 
> .
> For more options, visit https://groups.google.com/d/optout.
>
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Re: [SR-Users] RTPProxy issue?

[Maxim Sobolev]

Igor, yes, I'd say give 2.0 a try and see if the problem is still there.
There were tons of changes, particularly in the rtp_resize subsystem.

Thanks!


On Sun, Mar 8, 2015 at 7:31 AM, Igor Potjevlesch  wrote:

> Hello Maxim,
>
>
>
> I'm running legacy 1.2 or 1.4, not sure.
>
> I see in the latest code that the function is still there. Do you suggest
> to upgrade or there's a patch to make?
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *De la
> part de* Maxim Sobolev
> *Envoyé :* samedi 7 mars 2015 09:14
>
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
> Ah, ok, I see now. I did not realize you guys are using resizer. Which
> version of the software are you actually using? I.e. is it latest rel_2_0 /
> master, or some legacy 1.x code? We've done quite some revamping down
> there, so that it needs to be checked against the very latest code to make
> sure. Let us know.
>
> Thanks!
>
> On Mar 6, 2015 12:31 AM, "Igor Potjevlesch" 
> wrote:
>
> Hi Maxim,
>
>
>
> Hard to do because it's in production.
>
> I have a serious finding since yesterday on how this happened.
>
>
>
> My understanding is that the function "ts_less" returns FALSE into
> "rtp_resizer.c" because the timestamp between the two packets is > (1 <<
> 31) [for example: 3740425320].
>
> That's result in a drop of any following packets as I can see it into a
> capture.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *De la
> part de* Maxim Sobolev
> *Envoyé :* vendredi 6 mars 2015 07:44
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
> Hi Igor, that's bit strange, since the rtpproxy is not checking any of the
> rtp flags including marker bit. It would help if you can post a tcpdump
> capture of the streams in question along with the log output of the
> rtpproxy running at the "dbug" level. Thanks!
>
> On Mar 5, 2015 5:54 AM, "Igor Potjevlesch" 
> wrote:
>
> I reviewed again a call trace and I can be more precise: a RTP packet
> comes with a new SSRC and the Marker bit set to "True". This packet is
> properly forwarded.
>
>
>
> Then, just after this packet, another RTP packet containing a new SSRC
> with the huge timestamp and the Marker bit set to "True" is coming from the
> UA.
>
> The RTPProxy stops forward since this packet.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* Igor Potjevlesch [mailto:igor.potjevle...@gmail.com]
> *Envoyé :* jeudi 5 mars 2015 11:34
> *À :* mico...@gmail.com; 'Kamailio (SER) - Users Mailing List'
> *Objet :* RE: [SR-Users] RTPProxy issue?
>
>
>
> Hello,
>
>
>
> Thank you.
>
>
>
> Just to let you know, the RTPProxy is running in bridging mode.
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org
> ] *De la part de*
> Daniel-Constantin Mierla
> *Envoyé :* jeudi 5 mars 2015 09:33
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
> Hello,
>
> maybe Maxim (cc-ed) will be able to provide more insights.
>
> Cheers,
> DAniel
>
> On 04/03/15 16:59, Igor Potjevlesch wrote:
>
> Hello,
>
>
>
> I discovered an issue related to the handling of "timestamp" and/or
> "Marker bit" with rtpproxy (I use the latest Extension 20081224).
>
>
>
> The call-flow is the following: one UA places a call to A and put this
> call on hold. Then, the same UA call another number B. Individual streams
> are ok.
>
> When the UA tries to transfer A with B, the RTPProxy receive a RTP packet
> with a huge timestamp and the Marker bit set to "True".
>
>
>
> Just after this RTP packet, RTPProxy stop forward the RTP packets from A
> to B. B to C is still working.
>
>
>
> Anyone have an idea?
>
> Regards,
>
>
>
> Igor.
>
>
>
> ___
>
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>
> sr-users@lists.sip-router.org
>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
>
> Daniel-Constantin Mierla
>
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
> Kamailio World Conference, May 27-29, 2015
>
> Berlin, Germany - http://www.kama

Re: [SR-Users] rtpproxy failed to load

[Daniel-Constantin Mierla]

Hello,

use rtpproxy_manage() instead of force_rtp_proxy() -- that function was
replaced in newer versions.

Cheers,
Daniel

On 06/03/15 11:53, Tom Braarup Cuykens wrote:
> kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
>
> I found this guide for an older version of Kamailio, but there seems
> to be some issues when trying to use it on the latest stable version.
> When I try to start Kamailio i get this :
>
> [] 0(17153) INFO: tls [tls_init.c:399]: init_tls_compression():
> tls: init_tls: disabling compression... 0(17153) ERROR: 
> [cfg.y:3286]: yyparse(): cfg. parser: failed to find command
> force_rtp_proxy (params 0) 0(17153) :  [cfg.y:3426]:
> yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg,
> line 729, column 19: unknown command, missing loadmodule? 0(17153)
> ERROR:  [cfg.y:3286]: yyparse(): cfg. parser: failed to find
> command force_rtp_proxy (params 0) 0(17153) :  [cfg.y:3426]:
> yyerror_at(): parse error in config fil[FAILc/kamailio/kamailio.cfg,
> line 806, column 19: unknown command, missing loadmodule? ERROR: bad
> config file (2 errors) ...
>
> It seems it cannot load the rtpproxy module or that force_rtp_proxy
> command does not exist. For Debian the modules are located in
> /usr/lib/x86_64-linux-gnu/kamailio/modules/ and it was found there,
> seems to be loaded in the config file (line 5 and 235). According to
> the rtpproxy doc the module has this command.
>
> Does anyone have an idea what could be the issue ?
>
>
> I am planning to update this guide for the current version once I
> fixed all the issues.
> The most easy issues are the db/usernames/password and the path to the
> modules.
>
>

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com


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Re: [SR-Users] RTPProxy issue?

[Igor Potjevlesch]

Hello Maxim,

 

I'm running legacy 1.2 or 1.4, not sure.

I see in the latest code that the function is still there. Do you suggest to 
upgrade or there's a patch to make?

 

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de 
Maxim Sobolev
Envoyé : samedi 7 mars 2015 09:14
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Ah, ok, I see now. I did not realize you guys are using resizer. Which version 
of the software are you actually using? I.e. is it latest rel_2_0 / master, or 
some legacy 1.x code? We've done quite some revamping down there, so that it 
needs to be checked against the very latest code to make sure. Let us know.

Thanks!

On Mar 6, 2015 12:31 AM, "Igor Potjevlesch" mailto:igor.potjevle...@gmail.com> > wrote:

Hi Maxim,

 

Hard to do because it's in production.

I have a serious finding since yesterday on how this happened. 

 

My understanding is that the function "ts_less" returns FALSE into 
"rtp_resizer.c" because the timestamp between the two packets is > (1 << 31) 
[for example: 3740425320].

That's result in a drop of any following packets as I can see it into a capture.

 

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org 
<mailto:sr-users-boun...@lists.sip-router.org> ] De la part de Maxim Sobolev
Envoyé : vendredi 6 mars 2015 07:44
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Hi Igor, that's bit strange, since the rtpproxy is not checking any of the rtp 
flags including marker bit. It would help if you can post a tcpdump capture of 
the streams in question along with the log output of the rtpproxy running at 
the "dbug" level. Thanks!

On Mar 5, 2015 5:54 AM, "Igor Potjevlesch" mailto:igor.potjevle...@gmail.com> > wrote:

I reviewed again a call trace and I can be more precise: a RTP packet comes 
with a new SSRC and the Marker bit set to "True". This packet is properly 
forwarded.

 

Then, just after this packet, another RTP packet containing a new SSRC with the 
huge timestamp and the Marker bit set to "True" is coming from the UA. 

The RTPProxy stops forward since this packet.

 

Regards,

 

Igor.

 

De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com 
<mailto:igor.potjevle...@gmail.com> ] 
Envoyé : jeudi 5 mars 2015 11:34
À : mico...@gmail.com <mailto:mico...@gmail.com> ; 'Kamailio (SER) - Users 
Mailing List'
Objet : RE: [SR-Users] RTPProxy issue?

 

Hello,

 

Thank you.

 

Just to let you know, the RTPProxy is running in bridging mode.

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de 
Daniel-Constantin Mierla
Envoyé : jeudi 5 mars 2015 09:33
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Hello,

maybe Maxim (cc-ed) will be able to provide more insights.

Cheers,
DAniel

On 04/03/15 16:59, Igor Potjevlesch wrote:

Hello,

 

I discovered an issue related to the handling of "timestamp" and/or "Marker 
bit" with rtpproxy (I use the latest Extension 20081224).

 

The call-flow is the following: one UA places a call to A and put this call on 
hold. Then, the same UA call another number B. Individual streams are ok.

When the UA tries to transfer A with B, the RTPProxy receive a RTP packet with 
a huge timestamp and the Marker bit set to "True".

 

Just after this RTP packet, RTPProxy stop forward the RTP packets from A to B. 
B to C is still working.

 

Anyone have an idea?

Regards,

 

Igor.

 

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-- 
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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com


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Re: [SR-Users] RTPProxy issue?

[Maxim Sobolev]

Ah, ok, I see now. I did not realize you guys are using resizer. Which
version of the software are you actually using? I.e. is it latest rel_2_0 /
master, or some legacy 1.x code? We've done quite some revamping down
there, so that it needs to be checked against the very latest code to make
sure. Let us know.

Thanks!
On Mar 6, 2015 12:31 AM, "Igor Potjevlesch" 
wrote:

> Hi Maxim,
>
>
>
> Hard to do because it's in production.
>
> I have a serious finding since yesterday on how this happened.
>
>
>
> My understanding is that the function "ts_less" returns FALSE into
> "rtp_resizer.c" because the timestamp between the two packets is > (1 <<
> 31) [for example: 3740425320].
>
> That's result in a drop of any following packets as I can see it into a
> capture.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *De la
> part de* Maxim Sobolev
> *Envoyé :* vendredi 6 mars 2015 07:44
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
> Hi Igor, that's bit strange, since the rtpproxy is not checking any of the
> rtp flags including marker bit. It would help if you can post a tcpdump
> capture of the streams in question along with the log output of the
> rtpproxy running at the "dbug" level. Thanks!
>
> On Mar 5, 2015 5:54 AM, "Igor Potjevlesch" 
> wrote:
>
> I reviewed again a call trace and I can be more precise: a RTP packet
> comes with a new SSRC and the Marker bit set to "True". This packet is
> properly forwarded.
>
>
>
> Then, just after this packet, another RTP packet containing a new SSRC
> with the huge timestamp and the Marker bit set to "True" is coming from the
> UA.
>
> The RTPProxy stops forward since this packet.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* Igor Potjevlesch [mailto:igor.potjevle...@gmail.com]
> *Envoyé :* jeudi 5 mars 2015 11:34
> *À :* mico...@gmail.com; 'Kamailio (SER) - Users Mailing List'
> *Objet :* RE: [SR-Users] RTPProxy issue?
>
>
>
> Hello,
>
>
>
> Thank you.
>
>
>
> Just to let you know, the RTPProxy is running in bridging mode.
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org
> ] *De la part de*
> Daniel-Constantin Mierla
> *Envoyé :* jeudi 5 mars 2015 09:33
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
> Hello,
>
> maybe Maxim (cc-ed) will be able to provide more insights.
>
> Cheers,
> DAniel
>
> On 04/03/15 16:59, Igor Potjevlesch wrote:
>
> Hello,
>
>
>
> I discovered an issue related to the handling of "timestamp" and/or
> "Marker bit" with rtpproxy (I use the latest Extension 20081224).
>
>
>
> The call-flow is the following: one UA places a call to A and put this
> call on hold. Then, the same UA call another number B. Individual streams
> are ok.
>
> When the UA tries to transfer A with B, the RTPProxy receive a RTP packet
> with a huge timestamp and the Marker bit set to "True".
>
>
>
> Just after this RTP packet, RTPProxy stop forward the RTP packets from A
> to B. B to C is still working.
>
>
>
> Anyone have an idea?
>
> Regards,
>
>
>
> Igor.
>
>
>
> ___
>
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>
> sr-users@lists.sip-router.org
>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
>
> Daniel-Constantin Mierla
>
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
> Kamailio World Conference, May 27-29, 2015
>
> Berlin, Germany - http://www.kamailioworld.com
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> ___
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> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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Re: [SR-Users] RTPProxy issue?

[Igor Potjevlesch]

Hi Maxim,

 

Hard to do because it's in production.

I have a serious finding since yesterday on how this happened. 

 

My understanding is that the function "ts_less" returns FALSE into 
"rtp_resizer.c" because the timestamp between the two packets is > (1 << 31) 
[for example: 3740425320].

That's result in a drop of any following packets as I can see it into a capture.

 

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de 
Maxim Sobolev
Envoyé : vendredi 6 mars 2015 07:44
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Hi Igor, that's bit strange, since the rtpproxy is not checking any of the rtp 
flags including marker bit. It would help if you can post a tcpdump capture of 
the streams in question along with the log output of the rtpproxy running at 
the "dbug" level. Thanks!

On Mar 5, 2015 5:54 AM, "Igor Potjevlesch" mailto:igor.potjevle...@gmail.com> > wrote:

I reviewed again a call trace and I can be more precise: a RTP packet comes 
with a new SSRC and the Marker bit set to "True". This packet is properly 
forwarded.

 

Then, just after this packet, another RTP packet containing a new SSRC with the 
huge timestamp and the Marker bit set to "True" is coming from the UA. 

The RTPProxy stops forward since this packet.

 

Regards,

 

Igor.

 

De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com 
<mailto:igor.potjevle...@gmail.com> ] 
Envoyé : jeudi 5 mars 2015 11:34
À : mico...@gmail.com <mailto:mico...@gmail.com> ; 'Kamailio (SER) - Users 
Mailing List'
Objet : RE: [SR-Users] RTPProxy issue?

 

Hello,

 

Thank you.

 

Just to let you know, the RTPProxy is running in bridging mode.

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de 
Daniel-Constantin Mierla
Envoyé : jeudi 5 mars 2015 09:33
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Hello,

maybe Maxim (cc-ed) will be able to provide more insights.

Cheers,
DAniel

On 04/03/15 16:59, Igor Potjevlesch wrote:

Hello,

 

I discovered an issue related to the handling of "timestamp" and/or "Marker 
bit" with rtpproxy (I use the latest Extension 20081224).

 

The call-flow is the following: one UA places a call to A and put this call on 
hold. Then, the same UA call another number B. Individual streams are ok.

When the UA tries to transfer A with B, the RTPProxy receive a RTP packet with 
a huge timestamp and the Marker bit set to "True".

 

Just after this RTP packet, RTPProxy stop forward the RTP packets from A to B. 
B to C is still working.

 

Anyone have an idea?

Regards,

 

Igor.

 

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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com


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Re: [SR-Users] RTPProxy issue?

[Maxim Sobolev]

Hi Igor, that's bit strange, since the rtpproxy is not checking any of the
rtp flags including marker bit. It would help if you can post a tcpdump
capture of the streams in question along with the log output of the
rtpproxy running at the "dbug" level. Thanks!
On Mar 5, 2015 5:54 AM, "Igor Potjevlesch" 
wrote:

> I reviewed again a call trace and I can be more precise: a RTP packet
> comes with a new SSRC and the Marker bit set to "True". This packet is
> properly forwarded.
>
>
>
> Then, just after this packet, another RTP packet containing a new SSRC
> with the huge timestamp and the Marker bit set to "True" is coming from the
> UA.
>
> The RTPProxy stops forward since this packet.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* Igor Potjevlesch [mailto:igor.potjevle...@gmail.com]
> *Envoyé :* jeudi 5 mars 2015 11:34
> *À :* mico...@gmail.com; 'Kamailio (SER) - Users Mailing List'
> *Objet :* RE: [SR-Users] RTPProxy issue?
>
>
>
> Hello,
>
>
>
> Thank you.
>
>
>
> Just to let you know, the RTPProxy is running in bridging mode.
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org
> ] *De la part de*
> Daniel-Constantin Mierla
> *Envoyé :* jeudi 5 mars 2015 09:33
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
> Hello,
>
> maybe Maxim (cc-ed) will be able to provide more insights.
>
> Cheers,
> DAniel
>
> On 04/03/15 16:59, Igor Potjevlesch wrote:
>
> Hello,
>
>
>
> I discovered an issue related to the handling of "timestamp" and/or
> "Marker bit" with rtpproxy (I use the latest Extension 20081224).
>
>
>
> The call-flow is the following: one UA places a call to A and put this
> call on hold. Then, the same UA call another number B. Individual streams
> are ok.
>
> When the UA tries to transfer A with B, the RTPProxy receive a RTP packet
> with a huge timestamp and the Marker bit set to "True".
>
>
>
> Just after this RTP packet, RTPProxy stop forward the RTP packets from A
> to B. B to C is still working.
>
>
>
> Anyone have an idea?
>
> Regards,
>
>
>
> Igor.
>
>
>
> ___
>
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>
> sr-users@lists.sip-router.org
>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
>
> Daniel-Constantin Mierla
>
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
> Kamailio World Conference, May 27-29, 2015
>
> Berlin, Germany - http://www.kamailioworld.com
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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Re: [SR-Users] RTPProxy issue?

[Igor Potjevlesch]

I reviewed again a call trace and I can be more precise: a RTP packet comes
with a new SSRC and the Marker bit set to "True". This packet is properly
forwarded.

 

Then, just after this packet, another RTP packet containing a new SSRC with
the huge timestamp and the Marker bit set to "True" is coming from the UA. 

The RTPProxy stops forward since this packet.

 

Regards,

 

Igor.

 

De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] 
Envoyé : jeudi 5 mars 2015 11:34
À : mico...@gmail.com; 'Kamailio (SER) - Users Mailing List'
Objet : RE: [SR-Users] RTPProxy issue?

 

Hello,

 

Thank you.

 

Just to let you know, the RTPProxy is running in bridging mode.

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de
Daniel-Constantin Mierla
Envoyé : jeudi 5 mars 2015 09:33
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Hello,

maybe Maxim (cc-ed) will be able to provide more insights.

Cheers,
DAniel

On 04/03/15 16:59, Igor Potjevlesch wrote:

Hello,

 

I discovered an issue related to the handling of "timestamp" and/or "Marker
bit" with rtpproxy (I use the latest Extension 20081224).

 

The call-flow is the following: one UA places a call to A and put this call
on hold. Then, the same UA call another number B. Individual streams are ok.

When the UA tries to transfer A with B, the RTPProxy receive a RTP packet
with a huge timestamp and the Marker bit set to "True".

 

Just after this RTP packet, RTPProxy stop forward the RTP packets from A to
B. B to C is still working.

 

Anyone have an idea?

Regards,

 

Igor.





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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
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Re: [SR-Users] RTPProxy issue?

[Igor Potjevlesch]

Hello,

 

Thank you.

 

Just to let you know, the RTPProxy is running in bridging mode.

Regards,

 

Igor.

 

De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de
Daniel-Constantin Mierla
Envoyé : jeudi 5 mars 2015 09:33
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?

 

Hello,

maybe Maxim (cc-ed) will be able to provide more insights.

Cheers,
DAniel

On 04/03/15 16:59, Igor Potjevlesch wrote:

Hello,

 

I discovered an issue related to the handling of "timestamp" and/or "Marker
bit" with rtpproxy (I use the latest Extension 20081224).

 

The call-flow is the following: one UA places a call to A and put this call
on hold. Then, the same UA call another number B. Individual streams are ok.

When the UA tries to transfer A with B, the RTPProxy receive a RTP packet
with a huge timestamp and the Marker bit set to "True".

 

Just after this RTP packet, RTPProxy stop forward the RTP packets from A to
B. B to C is still working.

 

Anyone have an idea?

Regards,

 

Igor.






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-- 
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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
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Re: [SR-Users] RTPProxy issue?

[Daniel-Constantin Mierla]

Hello,

maybe Maxim (cc-ed) will be able to provide more insights.

Cheers,
DAniel

On 04/03/15 16:59, Igor Potjevlesch wrote:
>
> Hello,
>
>  
>
> I discovered an issue related to the handling of "timestamp" and/or
> "Marker bit" with rtpproxy (I use the latest Extension 20081224).
>
>  
>
> The call-flow is the following: one UA places a call to A and put this
> call on hold. Then, the same UA call another number B. Individual
> streams are ok.
>
> When the UA tries to transfer A with B, the RTPProxy receive a RTP
> packet with a huge timestamp and the Marker bit set to "True".
>
>  
>
> Just after this RTP packet, RTPProxy stop forward the RTP packets from
> A to B. B to C is still working.
>
>  
>
> Anyone have an idea?
>
> Regards,
>
>  
>
> Igor.
>
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com

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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[Frank Carmickle]

On Feb 16, 2015, at 7:27 PM, Ovidiu Sas  wrote:

> You could simply let the RTP traffic to flow directly between FS and
> endpoints (no need for rtpproxy).
> All you need to do is:
> - forward the appropriate RTP ports to FS;
> - fix the private IP in SDP by replacing it with the public IP for
> the inbound rtp streams (to FS).
> 
FS will do this for you if you set the ext-rtp-ip on the profile.


—FC


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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[asterisk asterisk]

Could you show the revelant codes in rtpproxy and kamailio.cfg? I am unable
to get the audio pass through from extranet to intranet as private IP
address is used after rtpproxy.

I use Kamailio 4.2 and rtpproxy in Debian wheezy. Both are installed from
repository.

On Tue, Feb 17, 2015 at 8:27 AM, Ovidiu Sas  wrote:

> You could simply let the RTP traffic to flow directly between FS and
> endpoints (no need for rtpproxy).
> All you need to do is:
>  - forward the appropriate RTP ports to FS;
>  - fix the private IP in SDP by replacing it with the public IP for
> the inbound rtp streams (to FS).
>
> -ovidiu
>
> On Mon, Feb 16, 2015 at 11:30 AM, Giovanni Maruzzelli 
> wrote:
> > dear Kamailians,
> >
> > I have Kamailio+rtpproxy in front of FreeSWITCH.
> >
> > Kamailio and FreeSWITCH are on the same private network.
> > Public Internet IP address ports are redirected to Kamailio and
> > rtpproxy (same situation as in Amazon EC2).
> > Clients comes from Internet, and make calls to Internet, SIP signaling
> > passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH
> > originate an outbound B leg INVITE, and then bridge the legs).
> >
> > Using rtpproxy with -A "advertise" patch from Daniel, this topology
> > works fine in a "traditional" telco way: rtp goes from caller to
> > rtpproxy to callee, and viceversa.
> >
> > Now I want to maintain FreeSWITCH in the middle of rtp flow all the
> > time, in a pure b2bua way, so it can control and analyze the media
> > streams.
> >
> > So, I need rtpproxy to act paying attention to direction, as in
> > caller->rtpproxy->freeswitch->rtpproxy->callee (and viceversa).
> >
> > Normally I would use Kamailio multihomed and rtpproxy in bridging
> > mode. But I cannot have a NIC on the public address.
> >
> > How I can use the "ie" "ei" feature of rtpproxy in an Amazon-EC2 like
> > environment? (eg: no public address attached to machine, but ports
> > redirection from public address).
> >
> > I read this trick from Hugh Waite:
> >
> > "I have used rtpproxy (with the advertised address patch) in Amazon to
> > bridge media between internet facing and private subnets in a VPC.
> > I found that I couldn’t use different advertised addresses depending
> > on which direction the signalling was going on a single private IP
> > address. I worked around this by allocating a second private ip
> > address to the instance and used that in the ‘bridge’.
> > -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15"
> >
> > Can you explain how to use this trick, or another way (without
> > additional addresses is gladly accepted!) to reach the same result
> > (rtp always passing through FreeSWITCH) ?
> >
> > Thank you all in advance,
> >
> > -giovanni
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> > ___
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> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[Ovidiu Sas]

You could simply let the RTP traffic to flow directly between FS and
endpoints (no need for rtpproxy).
All you need to do is:
 - forward the appropriate RTP ports to FS;
 - fix the private IP in SDP by replacing it with the public IP for
the inbound rtp streams (to FS).

-ovidiu

On Mon, Feb 16, 2015 at 11:30 AM, Giovanni Maruzzelli  wrote:
> dear Kamailians,
>
> I have Kamailio+rtpproxy in front of FreeSWITCH.
>
> Kamailio and FreeSWITCH are on the same private network.
> Public Internet IP address ports are redirected to Kamailio and
> rtpproxy (same situation as in Amazon EC2).
> Clients comes from Internet, and make calls to Internet, SIP signaling
> passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH
> originate an outbound B leg INVITE, and then bridge the legs).
>
> Using rtpproxy with -A "advertise" patch from Daniel, this topology
> works fine in a "traditional" telco way: rtp goes from caller to
> rtpproxy to callee, and viceversa.
>
> Now I want to maintain FreeSWITCH in the middle of rtp flow all the
> time, in a pure b2bua way, so it can control and analyze the media
> streams.
>
> So, I need rtpproxy to act paying attention to direction, as in
> caller->rtpproxy->freeswitch->rtpproxy->callee (and viceversa).
>
> Normally I would use Kamailio multihomed and rtpproxy in bridging
> mode. But I cannot have a NIC on the public address.
>
> How I can use the "ie" "ei" feature of rtpproxy in an Amazon-EC2 like
> environment? (eg: no public address attached to machine, but ports
> redirection from public address).
>
> I read this trick from Hugh Waite:
>
> "I have used rtpproxy (with the advertised address patch) in Amazon to
> bridge media between internet facing and private subnets in a VPC.
> I found that I couldn’t use different advertised addresses depending
> on which direction the signalling was going on a single private IP
> address. I worked around this by allocating a second private ip
> address to the instance and used that in the ‘bridge’.
> -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15"
>
> Can you explain how to use this trick, or another way (without
> additional addresses is gladly accepted!) to reach the same result
> (rtp always passing through FreeSWITCH) ?
>
> Thank you all in advance,
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[Richard Fuchs]

On 16/02/15 01:12 PM, Virmantas Variakojis wrote:
> Could you provide us a little example? For examlple i have kamailio with
> three interfaces: two interfaces (vlan's look at two different
> providers) and third interface looks at sip clients.

You would define two interfaces with different names, for example
--interface=public/10.0.1.15!54.86.X.X for outside media and
--interface=local/10.0.1.15 for local media. You would then use two
direction=... options in the offer to determine where A side and B side
are located, respectively. You can also call the interfaces "external"
and "internal" and use these flags instead, which mirrors rtpproxy's
behaviour.

Cheers

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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[Virmantas Variakojis]

Could you provide us a little example? For examlple i have kamailio with
three interfaces: two interfaces (vlan's look at two different providers)
and third interface looks at sip clients.
Thank's in advance!
2015 vas. 16 20:04 "Richard Fuchs"  rašė:

> On 16/02/15 01:00 PM, Virmantas Variakojis wrote:
> > Hi,
> >
> > There pathch with -A can be found or it is allready implemented into
> > specific rtpengine version?
>
> Latest master from git. The command line syntax is a bit different from
> rtpproxy, but the basic idea is the same.
>
> Cheers
>
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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[Richard Fuchs]

On 16/02/15 01:00 PM, Virmantas Variakojis wrote:
> Hi,
> 
> There pathch with -A can be found or it is allready implemented into
> specific rtpengine version?

Latest master from git. The command line syntax is a bit different from
rtpproxy, but the basic idea is the same.

Cheers

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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[Virmantas Variakojis]

Hi,

There pathch with -A can be found or it is allready implemented into
specific rtpengine version?
2015 vas. 16 19:50 "Richard Fuchs"  rašė:

> On 16/02/15 12:39 PM, Muhammad Shahzad wrote:
> > I haven't done something like that myself but i think if you use
> > RTPEngine with "media-address" set correctly in offer and answer
> > functions, you can easily achieve this. Simply check if request/reply is
> > coming from FS or the end-user and adjust the media appropriately
> > without even invoking auto-bridge etc.
>
> Or perhaps with multiple interface specifications, binding to the same
> local address but with different advertised addresses, as suggested in
> the OP.
>
> Cheers
>
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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[Richard Fuchs]

On 16/02/15 12:39 PM, Muhammad Shahzad wrote:
> I haven't done something like that myself but i think if you use
> RTPEngine with "media-address" set correctly in offer and answer
> functions, you can easily achieve this. Simply check if request/reply is
> coming from FS or the end-user and adjust the media appropriately
> without even invoking auto-bridge etc.

Or perhaps with multiple interface specifications, binding to the same
local address but with different advertised addresses, as suggested in
the OP.

Cheers

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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[Muhammad Shahzad]

BTW, if nothing works, you can always use "network:msg" event route to find
/ replace any part of the SIP request and response, including media IP in
SDP. ;-)

http://kamailio.org/docs/modules/4.2.x/modules/corex.html#async.evr.network_io

Thank you.



On Mon, Feb 16, 2015 at 6:39 PM, Muhammad Shahzad 
wrote:

> I haven't done something like that myself but i think if you use RTPEngine
> with "media-address" set correctly in offer and answer functions, you can
> easily achieve this. Simply check if request/reply is coming from FS or the
> end-user and adjust the media appropriately without even invoking
> auto-bridge etc.
>
> Thank you.
>
>
>
> On Mon, Feb 16, 2015 at 5:30 PM, Giovanni Maruzzelli 
> wrote:
>
>> dear Kamailians,
>>
>> I have Kamailio+rtpproxy in front of FreeSWITCH.
>>
>> Kamailio and FreeSWITCH are on the same private network.
>> Public Internet IP address ports are redirected to Kamailio and
>> rtpproxy (same situation as in Amazon EC2).
>> Clients comes from Internet, and make calls to Internet, SIP signaling
>> passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH
>> originate an outbound B leg INVITE, and then bridge the legs).
>>
>> Using rtpproxy with -A "advertise" patch from Daniel, this topology
>> works fine in a "traditional" telco way: rtp goes from caller to
>> rtpproxy to callee, and viceversa.
>>
>> Now I want to maintain FreeSWITCH in the middle of rtp flow all the
>> time, in a pure b2bua way, so it can control and analyze the media
>> streams.
>>
>> So, I need rtpproxy to act paying attention to direction, as in
>> caller->rtpproxy->freeswitch->rtpproxy->callee (and viceversa).
>>
>> Normally I would use Kamailio multihomed and rtpproxy in bridging
>> mode. But I cannot have a NIC on the public address.
>>
>> How I can use the "ie" "ei" feature of rtpproxy in an Amazon-EC2 like
>> environment? (eg: no public address attached to machine, but ports
>> redirection from public address).
>>
>> I read this trick from Hugh Waite:
>>
>> "I have used rtpproxy (with the advertised address patch) in Amazon to
>> bridge media between internet facing and private subnets in a VPC.
>> I found that I couldn’t use different advertised addresses depending
>> on which direction the signalling was going on a single private IP
>> address. I worked around this by allocating a second private ip
>> address to the instance and used that in the ‘bridge’.
>> -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15"
>>
>> Can you explain how to use this trick, or another way (without
>> additional addresses is gladly accepted!) to reach the same result
>> (rtp always passing through FreeSWITCH) ?
>>
>> Thank you all in advance,
>>
>> -giovanni
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> ___
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>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[Muhammad Shahzad]

I haven't done something like that myself but i think if you use RTPEngine
with "media-address" set correctly in offer and answer functions, you can
easily achieve this. Simply check if request/reply is coming from FS or the
end-user and adjust the media appropriately without even invoking
auto-bridge etc.

Thank you.



On Mon, Feb 16, 2015 at 5:30 PM, Giovanni Maruzzelli 
wrote:

> dear Kamailians,
>
> I have Kamailio+rtpproxy in front of FreeSWITCH.
>
> Kamailio and FreeSWITCH are on the same private network.
> Public Internet IP address ports are redirected to Kamailio and
> rtpproxy (same situation as in Amazon EC2).
> Clients comes from Internet, and make calls to Internet, SIP signaling
> passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH
> originate an outbound B leg INVITE, and then bridge the legs).
>
> Using rtpproxy with -A "advertise" patch from Daniel, this topology
> works fine in a "traditional" telco way: rtp goes from caller to
> rtpproxy to callee, and viceversa.
>
> Now I want to maintain FreeSWITCH in the middle of rtp flow all the
> time, in a pure b2bua way, so it can control and analyze the media
> streams.
>
> So, I need rtpproxy to act paying attention to direction, as in
> caller->rtpproxy->freeswitch->rtpproxy->callee (and viceversa).
>
> Normally I would use Kamailio multihomed and rtpproxy in bridging
> mode. But I cannot have a NIC on the public address.
>
> How I can use the "ie" "ei" feature of rtpproxy in an Amazon-EC2 like
> environment? (eg: no public address attached to machine, but ports
> redirection from public address).
>
> I read this trick from Hugh Waite:
>
> "I have used rtpproxy (with the advertised address patch) in Amazon to
> bridge media between internet facing and private subnets in a VPC.
> I found that I couldn’t use different advertised addresses depending
> on which direction the signalling was going on a single private IP
> address. I worked around this by allocating a second private ip
> address to the instance and used that in the ‘bridge’.
> -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15"
>
> Can you explain how to use this trick, or another way (without
> additional addresses is gladly accepted!) to reach the same result
> (rtp always passing through FreeSWITCH) ?
>
> Thank you all in advance,
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)

[Daniel-Constantin Mierla]

Hello,

rtpproxy doing bridging requires two network interfaces to work with.

You can try one of the following:
- let freeswitch advertise the public ip for media and skip rtpproxy
completely
- use the second parameter of rtpproxy_manage() to set the advertised ip
address for media and don't configure rtpproxy in bridge mode

Cheers,
Daniel

On 16/02/15 17:30, Giovanni Maruzzelli wrote:
> dear Kamailians,
>
> I have Kamailio+rtpproxy in front of FreeSWITCH.
>
> Kamailio and FreeSWITCH are on the same private network.
> Public Internet IP address ports are redirected to Kamailio and
> rtpproxy (same situation as in Amazon EC2).
> Clients comes from Internet, and make calls to Internet, SIP signaling
> passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH
> originate an outbound B leg INVITE, and then bridge the legs).
>
> Using rtpproxy with -A "advertise" patch from Daniel, this topology
> works fine in a "traditional" telco way: rtp goes from caller to
> rtpproxy to callee, and viceversa.
>
> Now I want to maintain FreeSWITCH in the middle of rtp flow all the
> time, in a pure b2bua way, so it can control and analyze the media
> streams.
>
> So, I need rtpproxy to act paying attention to direction, as in
> caller->rtpproxy->freeswitch->rtpproxy->callee (and viceversa).
>
> Normally I would use Kamailio multihomed and rtpproxy in bridging
> mode. But I cannot have a NIC on the public address.
>
> How I can use the "ie" "ei" feature of rtpproxy in an Amazon-EC2 like
> environment? (eg: no public address attached to machine, but ports
> redirection from public address).
>
> I read this trick from Hugh Waite:
>
> "I have used rtpproxy (with the advertised address patch) in Amazon to
> bridge media between internet facing and private subnets in a VPC.
> I found that I couldn’t use different advertised addresses depending
> on which direction the signalling was going on a single private IP
> address. I worked around this by allocating a second private ip
> address to the instance and used that in the ‘bridge’.
> -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15"
>
> Can you explain how to use this trick, or another way (without
> additional addresses is gladly accepted!) to reach the same result
> (rtp always passing through FreeSWITCH) ?
>
> Thank you all in advance,
>
> -giovanni
>

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com


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Re: [SR-Users] rtpproxy & extra_id_pv

[Richard Fuchs]

On 09/29/14 14:29, Marino Mileti wrote:
> Wow! Do you have an example of how to do that? How I have to modify my
> kamailio.conf in order to instructs rtpproxy to user from-tag & to-tag in
> this way?

You don't have to do anything, tags are already included in all the
messages.

cheers

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Re: [SR-Users] rtpproxy & extra_id_pv

[Richard Fuchs]

On 09/29/14 14:08, Marino Mileti wrote:
> But with from-tag and To-tag it's possible to instruct rtpengine to generate
> new couple of ports for each branch of a call? In the source code of
> rtpengine it seems that it check only the callid parameter

Yes it will. The call-id is only a vague umbrella under which the
from/to tags are collected. All media parameters, ports etc are
associated with the tags, and a single call can contain any number of
different from/to tags, all with their own separate parameters.

cheers

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Re: [SR-Users] rtpproxy & extra_id_pv

[Richard Fuchs]

On 09/26/14 16:57, Marino Mileti wrote:
> Hello,
> 
>> On Friday 26 September 2014 16:44:44 Marino Mileti wrote:
>>> Hi guys,
>>> I've seen that setting the parameter extra_id_pv, every branch should 
>>> be a different callid..
>>> How can i set this parameter? I've tried with :
>>> modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)")
>>>
>>> but in the INVITE message the callid is still the same for all 
>>> branches. Any suggest?
> 
>> Did you assign a value to $avp(extra) in the script, before calling any of
> the rtpproxy functions?
> 
>> Did you use the 'b' parameter in the call to rtpproxy_*() functions?
>> --
>> Alex Hermann
> 
> Yes, i've assigned a value in the script, just to try I've assigned $rU and
> on the log of rtpengine I can see the value of "via-branch" valorized
> correctly. I've also used the 'd' parameter in rtpproxy_manage() function.
> 
> I've also checked on the source code of rtpengine and the part regarding the
> "parse & get" of via-branch value is commented...I'm using the master branch
> of rtpengine.
> 
> This is the piece of code of call_interfaces.c where the via-branch
> parameter is commented:
> //bencode_dictionary_get_str(input, "via-branch", &viabranch);

The via-branch is currently unused in rtpengine, even though it does
appear in the control messages. From-tag and to-tag are sufficient to
identify a unique dialogue within a call.

cheers

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Re: [SR-Users] rtpproxy & extra_id_pv

[Alex Hermann]

On Friday 26 September 2014 16:44:44 Marino Mileti wrote:
> Hi guys,
> I've seen that setting the parameter extra_id_pv, every branch should be hae 
different callid..
> How can i set this parameter? I've tried with :
> modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)")
> 
> but in the INVITE message the callid is still the same for all branches. Any 
suggest?

Did you assign a value to $avp(extra) in the script, before calling any of the 
rtpproxy functions?

Did you use the 'b' parameter in the call to rtpproxy_*() functions?
-- 
Alex Hermann

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Re: [SR-Users] RTPPROXY & BRANCH

[Richard Fuchs]

On 23/09/14 04:35 AM, marino.mil...@alice.it wrote:


I've a problem with rtpproxy during a parallel ring scenario.
I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I
try to call them in parallel mode (ringall) rtpproxy module sends in to
the INVITE the same RTP ports.

Is it possible to manage rtpproxy in order to generate a couple of new
port for each branch?


Not at the moment. The logic is that one endpoint advertised by the 
client is mapped to one endpoint advertised by rtpengine. Why would you 
like a separate set of ports?


cheers

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Re: [SR-Users] RTPProxy timeout

[Carsten Bock]

Hi Igor,

Please check the rtpproxy.patch in the test-folder. It contains the
patch for RTPProxy.
i haven't used the enhancement myself for quite some time now
(replaced it with SEMS and rtpproxy-ng / rtpengine instead); but when
i wrote it, it  had to be XML-RPC.

Kind regards,
Carsten

2014-05-23 16:03 GMT+02:00 Igor Potjevlesch :
> Hi Carsten,
>
> Oh ok, I understand.
> Where can I find the patch as he is not in the test/ directory? I compiled 
> Kamailio from source, not GIT.
>
> Then, about the configuration, is it mandatory to use a XMLRPC interface or 
> that could be done through TCP or UDP? I saw in the man page  of RTPProxy 
> that the argument "-n" support tcp: for example.
>
> Regards,
>
> Igor.
>
>
> -Message d'origine-
> De : sr-users-boun...@lists.sip-router.org 
> [mailto:sr-users-boun...@lists.sip-router.org] De la part de Carsten Bock
> Envoyé : jeudi 22 mai 2014 20:08
> À : Kamailio (SER) - Users Mailing List
> Objet : Re: [SR-Users] RTPProxy timeout
>
> Hi Igor,
>
> it worked pretty well at the time i wrote the patch.
> I've sent the patch to the developers of RTPProxy, but i never received a 
> reply nor comment on the patch. That's why you have to apply the patch 
> yourself.
>
> Kind regards,
> Carsten
>
> 2014-05-22 19:32 GMT+02:00 Igor Potjevlesch :
>> Hi Carsten,
>>
>> Thank you for your answer.
>> I don't have this patch into test/ directory. Just a sample config of 
>> kamailio.cfg and 2 txt files with samples commands for rtpproxy and pjsua.
>>
>> I understand that finally you didn't succeed in patching your rtpproxy and 
>> use the timeout functionality?
>>
>> Regards,
>>
>> Igor.
>>
>> -Message d'origine-----
>> De : sr-users-boun...@lists.sip-router.org
>> [mailto:sr-users-boun...@lists.sip-router.org] De la part de Carsten
>> Bock Envoyé : jeudi 22 mai 2014 18:25 À : Kamailio (SER) - Users
>> Mailing List Objet : Re: [SR-Users] RTPProxy timeout
>>
>> Hi Igor,
>>
>> in order to use this feature, you need to apply a patch to RTPProxy.
>> The patch can be found in the "test" subdirectory in modules/rtpproxy/.
>>
>> I tried to get the patch into the rtpproxy, but i never got an answer from 
>> the developers of RTPProxy.
>>
>> Kind regards,
>> Carsten
>>
>> 2014-05-22 11:21 GMT+02:00 Igor Potjevlesch :
>>> Hello,
>>>
>>>
>>>
>>> I can’t figure out how the “timeout_socket” should be configured.
>>>
>>> I try with the default parameter in the doc:
>>> xmlrpc:http://127.0.0.1:8000/RPC2
>>>
>>> I launch RTPProxy like that:  -n tcp:127.0.0.1:8000
>>>
>>> RTPProxy socket is setup like: "rtpproxy_sock", "udp:127.0.0.1:7722".
>>>
>>>
>>>
>>> It looks like there is no communication on the port 8000. The goal is
>>> to setup a timeout in case of asymmetric RTP.
>>>
>>>
>>>
>>> Many help would be appreciate. Thank you!
>>>
>>>
>>>
>>> Regards,
>>>
>>>
>>>
>>> Igor.
>>>
>>>
>>> ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>> list sr-users@lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>>
>> --
>> Carsten Bock
>> CEO (Geschäftsführer)
>>
>> ng-voice GmbH
>> Schomburgstr. 80
>> D-22767 Hamburg / Germany
>>
>> http://www.ng-voice.com
>> mailto:cars...@ng-voice.com
>>
>> Office +49 40 34927219
>> Fax +49 40 34927220
>>
>> Sitz der Gesellschaft: Hamburg
>> Registergericht: Amtsgericht Hamburg, HRB 120189
>> Geschäftsführer: Carsten Bock
>> Ust-ID: DE279344284
>>
>> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
>> http://www.ng-voice.com/imprint/
>>
>> ___
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>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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>>
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>
>
>
> --
> Carsten Bock
> CEO (Geschäftsführer)
&

Re: [SR-Users] RTPProxy timeout

[Igor Potjevlesch]

Hi Carsten,

Oh ok, I understand.
Where can I find the patch as he is not in the test/ directory? I compiled 
Kamailio from source, not GIT.

Then, about the configuration, is it mandatory to use a XMLRPC interface or 
that could be done through TCP or UDP? I saw in the man page  of RTPProxy that 
the argument "-n" support tcp: for example.

Regards,

Igor.


-Message d'origine-
De : sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] De la part de Carsten Bock
Envoyé : jeudi 22 mai 2014 20:08
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy timeout

Hi Igor,

it worked pretty well at the time i wrote the patch.
I've sent the patch to the developers of RTPProxy, but i never received a reply 
nor comment on the patch. That's why you have to apply the patch yourself.

Kind regards,
Carsten

2014-05-22 19:32 GMT+02:00 Igor Potjevlesch :
> Hi Carsten,
>
> Thank you for your answer.
> I don't have this patch into test/ directory. Just a sample config of 
> kamailio.cfg and 2 txt files with samples commands for rtpproxy and pjsua.
>
> I understand that finally you didn't succeed in patching your rtpproxy and 
> use the timeout functionality?
>
> Regards,
>
> Igor.
>
> -Message d'origine-
> De : sr-users-boun...@lists.sip-router.org 
> [mailto:sr-users-boun...@lists.sip-router.org] De la part de Carsten 
> Bock Envoyé : jeudi 22 mai 2014 18:25 À : Kamailio (SER) - Users 
> Mailing List Objet : Re: [SR-Users] RTPProxy timeout
>
> Hi Igor,
>
> in order to use this feature, you need to apply a patch to RTPProxy.
> The patch can be found in the "test" subdirectory in modules/rtpproxy/.
>
> I tried to get the patch into the rtpproxy, but i never got an answer from 
> the developers of RTPProxy.
>
> Kind regards,
> Carsten
>
> 2014-05-22 11:21 GMT+02:00 Igor Potjevlesch :
>> Hello,
>>
>>
>>
>> I can’t figure out how the “timeout_socket” should be configured.
>>
>> I try with the default parameter in the doc:
>> xmlrpc:http://127.0.0.1:8000/RPC2
>>
>> I launch RTPProxy like that:  -n tcp:127.0.0.1:8000
>>
>> RTPProxy socket is setup like: "rtpproxy_sock", "udp:127.0.0.1:7722".
>>
>>
>>
>> It looks like there is no communication on the port 8000. The goal is 
>> to setup a timeout in case of asymmetric RTP.
>>
>>
>>
>> Many help would be appreciate. Thank you!
>>
>>
>>
>> Regards,
>>
>>
>>
>> Igor.
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>> list sr-users@lists.sip-router.org 
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> --
> Carsten Bock
> CEO (Geschäftsführer)
>
> ng-voice GmbH
> Schomburgstr. 80
> D-22767 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:cars...@ng-voice.com
>
> Office +49 40 34927219
> Fax +49 40 34927220
>
> Sitz der Gesellschaft: Hamburg
> Registergericht: Amtsgericht Hamburg, HRB 120189
> Geschäftsführer: Carsten Bock
> Ust-ID: DE279344284
>
> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
> http://www.ng-voice.com/imprint/
>
> ___
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



--
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CEO (Geschäftsführer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:cars...@ng-voice.com

Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
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Ust-ID: DE279344284

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Re: [SR-Users] RTPProxy timeout

[Carsten Bock]

Hi Igor,

it worked pretty well at the time i wrote the patch.
I've sent the patch to the developers of RTPProxy, but i never
received a reply nor comment on the patch. That's why you have to
apply the patch yourself.

Kind regards,
Carsten

2014-05-22 19:32 GMT+02:00 Igor Potjevlesch :
> Hi Carsten,
>
> Thank you for your answer.
> I don't have this patch into test/ directory. Just a sample config of 
> kamailio.cfg and 2 txt files with samples commands for rtpproxy and pjsua.
>
> I understand that finally you didn't succeed in patching your rtpproxy and 
> use the timeout functionality?
>
> Regards,
>
> Igor.
>
> -Message d'origine-
> De : sr-users-boun...@lists.sip-router.org 
> [mailto:sr-users-boun...@lists.sip-router.org] De la part de Carsten Bock
> Envoyé : jeudi 22 mai 2014 18:25
> À : Kamailio (SER) - Users Mailing List
> Objet : Re: [SR-Users] RTPProxy timeout
>
> Hi Igor,
>
> in order to use this feature, you need to apply a patch to RTPProxy.
> The patch can be found in the "test" subdirectory in modules/rtpproxy/.
>
> I tried to get the patch into the rtpproxy, but i never got an answer from 
> the developers of RTPProxy.
>
> Kind regards,
> Carsten
>
> 2014-05-22 11:21 GMT+02:00 Igor Potjevlesch :
>> Hello,
>>
>>
>>
>> I can’t figure out how the “timeout_socket” should be configured.
>>
>> I try with the default parameter in the doc:
>> xmlrpc:http://127.0.0.1:8000/RPC2
>>
>> I launch RTPProxy like that:  -n tcp:127.0.0.1:8000
>>
>> RTPProxy socket is setup like: "rtpproxy_sock", "udp:127.0.0.1:7722".
>>
>>
>>
>> It looks like there is no communication on the port 8000. The goal is
>> to setup a timeout in case of asymmetric RTP.
>>
>>
>>
>> Many help would be appreciate. Thank you!
>>
>>
>>
>> Regards,
>>
>>
>>
>> Igor.
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>> list sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> --
> Carsten Bock
> CEO (Geschäftsführer)
>
> ng-voice GmbH
> Schomburgstr. 80
> D-22767 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:cars...@ng-voice.com
>
> Office +49 40 34927219
> Fax +49 40 34927220
>
> Sitz der Gesellschaft: Hamburg
> Registergericht: Amtsgericht Hamburg, HRB 120189
> Geschäftsführer: Carsten Bock
> Ust-ID: DE279344284
>
> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
> http://www.ng-voice.com/imprint/
>
> ___
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> sr-users@lists.sip-router.org 
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



-- 
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CEO (Geschäftsführer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:cars...@ng-voice.com

Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284

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Re: [SR-Users] RTPProxy timeout

[Igor Potjevlesch]

Hi Carsten,

Thank you for your answer.
I don't have this patch into test/ directory. Just a sample config of 
kamailio.cfg and 2 txt files with samples commands for rtpproxy and pjsua.

I understand that finally you didn't succeed in patching your rtpproxy and use 
the timeout functionality?

Regards,

Igor.

-Message d'origine-
De : sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] De la part de Carsten Bock
Envoyé : jeudi 22 mai 2014 18:25
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy timeout

Hi Igor,

in order to use this feature, you need to apply a patch to RTPProxy.
The patch can be found in the "test" subdirectory in modules/rtpproxy/.

I tried to get the patch into the rtpproxy, but i never got an answer from the 
developers of RTPProxy.

Kind regards,
Carsten

2014-05-22 11:21 GMT+02:00 Igor Potjevlesch :
> Hello,
>
>
>
> I can’t figure out how the “timeout_socket” should be configured.
>
> I try with the default parameter in the doc:
> xmlrpc:http://127.0.0.1:8000/RPC2
>
> I launch RTPProxy like that:  -n tcp:127.0.0.1:8000
>
> RTPProxy socket is setup like: "rtpproxy_sock", "udp:127.0.0.1:7722".
>
>
>
> It looks like there is no communication on the port 8000. The goal is 
> to setup a timeout in case of asymmetric RTP.
>
>
>
> Many help would be appreciate. Thank you!
>
>
>
> Regards,
>
>
>
> Igor.
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
> list sr-users@lists.sip-router.org 
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



--
Carsten Bock
CEO (Geschäftsführer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:cars...@ng-voice.com

Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/

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Re: [SR-Users] RTPProxy timeout

[Carsten Bock]

Hi Igor,

in order to use this feature, you need to apply a patch to RTPProxy.
The patch can be found in the "test" subdirectory in modules/rtpproxy/.

I tried to get the patch into the rtpproxy, but i never got an answer
from the developers of RTPProxy.

Kind regards,
Carsten

2014-05-22 11:21 GMT+02:00 Igor Potjevlesch :
> Hello,
>
>
>
> I can’t figure out how the “timeout_socket” should be configured.
>
> I try with the default parameter in the doc:
> xmlrpc:http://127.0.0.1:8000/RPC2
>
> I launch RTPProxy like that:  -n tcp:127.0.0.1:8000
>
> RTPProxy socket is setup like: "rtpproxy_sock", "udp:127.0.0.1:7722".
>
>
>
> It looks like there is no communication on the port 8000. The goal is to
> setup a timeout in case of asymmetric RTP.
>
>
>
> Many help would be appreciate. Thank you!
>
>
>
> Regards,
>
>
>
> Igor.
>
>
> ___
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Carsten Bock
CEO (Geschäftsführer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:cars...@ng-voice.com

Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/

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Re: [SR-Users] RTPProxy and premature RTP

[Daniel Tryba]

On Wednesday 23 April 2014 13:41:42 Alex Balashov wrote:
> Just of curiosity, how exactly does the "havoc" manifest itself? What 
> are the symptoms of this being a problem?

Havoc might be a bit strong (though "great confusion and disorder" fits well). 
Bad sound quality (if any sound) for a longer period due to multiple jitter 
buffer overflows for that one channel and if the endpoint is on a limited 
circuit drops in the other channels as well. It manifests on a particular PBX 
(Swyx), most endpoints are Patton Smartnodes and they seem to cope well.

-- 

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Telefoon: 040 293 8661 - Fax: 040 293 8658
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Re: [SR-Users] RTPProxy and premature RTP

[Alex Balashov]

On 04/23/2014 07:36 AM, Daniel Tryba wrote:


I'm having some troubles with a provider sending RTP before a 183
Session Progress or 200 OK (I see up to 1s of rtp prematurely). The
machine is running rtpproxy and apparently rtpproxy buffers these rtp
packets and flushed them in one burst when the 183/200 arrive, this
creates havock in some endpoints but is undesirable in all cases IMHO
if the delay of the 182/200 is to high (>0.1s).


Just of curiosity, how exactly does the "havoc" manifest itself? What 
are the symptoms of this being a problem?


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Re: [SR-Users] RTPProxy/Mediaproxy issue

[Ravi]

Dear Daniel,

My whole set-up remains same in both the cases, only change is running
RTPproxy with kamailio server and without running RTPproxy server.
Now When i run RTPproxy with kamailio server, RTP packets has to reach up to
RTPproxy server to get relay to other end client right ? 
But how this client-to client RTP traffic gets transfer? i mean in
client-to-client method, is RTP packets are passing through only WAP (in my
above set-up) to reach other end client ?

Regards,
Ravi




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Re: [SR-Users] RTPProxy/Mediaproxy issue

[Daniel Grotti]

Hi,
When you don't use RTPproxy, RTP traffic is sent end-to-end.
What is the RTP path in case of client-to-client RTP ?
Is it the same ?

Daniel






On 02/25/2014 06:34 PM, Ravi wrote:
> Dear Daniel,
> 
> Thank you for the response,
> 
>> Do you have any router, firewall something between clients and mediaproxy ?
> 
> Ya I have Routers And switch in between clients and Rtpproxy, My set-up is
> as follows:
> 
>   SIP Clients<-->WAP+WiMax CPE <--> WiMax Base station <>
> RTPproxy+Kamailio server.
> 
> Both the SIP clients A and B are connected to Wi-Fi Access point (WAP).
> When i run RTPproxy server with kamailio server then only RTPproxy receiving
> very High RTP packet loss, But when only running kamailio server without
> RTPproxy server Then there is lesser RTP packet loss (< 5%).
> 
> Whats could be the reason for this behavior ?
> 
> How can i reduce this RTP packet loss ?
> 
> I hope you could help me.
> 
> Awaiting your reply,
> 
> Regards,
> Ravi
>  
> 
> 
> 
> 
> 
> --
> View this message in context: 
> http://sip-router.1086192.n5.nabble.com/RTPProxy-Mediaproxy-issue-tp125121p125351.html
> Sent from the Users mailing list archive at Nabble.com.
> 
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Re: [SR-Users] RTPProxy/Mediaproxy issue

[Ravi]

Dear Daniel,

Thank you for the response,

>Do you have any router, firewall something between clients and mediaproxy ?

Ya I have Routers And switch in between clients and Rtpproxy, My set-up is
as follows:

  SIP Clients<-->WAP+WiMax CPE <--> WiMax Base station <>
RTPproxy+Kamailio server.

Both the SIP clients A and B are connected to Wi-Fi Access point (WAP).
When i run RTPproxy server with kamailio server then only RTPproxy receiving
very High RTP packet loss, But when only running kamailio server without
RTPproxy server Then there is lesser RTP packet loss (< 5%).

Whats could be the reason for this behavior ?

How can i reduce this RTP packet loss ?

I hope you could help me.

Awaiting your reply,

Regards,
Ravi
 





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Re: [SR-Users] RTPProxy/Mediaproxy issue

[Daniel Grotti]

Hi Ravi,
if you media/rtp proxy is receiving such packet loss, it means that
something behind him is cutting the traffic off, somehow.
So you should investigate there, in their configuration.
If your analysis is right, it seems your loosing around 80% of your packets.

Do you have any router, firewall something between clients and mediaproxy ?

Daniel



On 02/24/2014 03:44 PM, Ravi wrote:
> Dear Daniel,
> Thank you again,
> 
> Ya i am investigating on this issue, by the way can you give any comments on
> my questions in the previous mail?  I just wanna clarify those things to
> rectify this packet loss issue. i googled about those questions but still
> ended with the same confusion status and didnt got any prompt information.
> And here below is the attachment, that shows RTP packet loss(using
> wireshark) in my set-up. Can you please suggest me how can i troubleshoot
> this RTP packet loss issue ?
> 
> And in that Attachment the IP addresses are like this:
> 192.168.2.235 and 192.168.2.239 are clients and 192.168.2.52  on which
> RTPproxy and Kamailio are running.
> 
> Awaiting your reply.
> 
> regards,
> Ravi
> RTP_Packet_loss.png
>   
> 
> 
> 
> 
> --
> View this message in context: 
> http://sip-router.1086192.n5.nabble.com/RTPProxy-Mediaproxy-issue-tp125121p125313.html
> Sent from the Users mailing list archive at Nabble.com.
> 
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Re: [SR-Users] RTPProxy/Mediaproxy issue

[Ravi]

Dear Daniel,
Thank you again,

Ya i am investigating on this issue, by the way can you give any comments on
my questions in the previous mail?  I just wanna clarify those things to
rectify this packet loss issue. i googled about those questions but still
ended with the same confusion status and didnt got any prompt information.
And here below is the attachment, that shows RTP packet loss(using
wireshark) in my set-up. Can you please suggest me how can i troubleshoot
this RTP packet loss issue ?

And in that Attachment the IP addresses are like this:
192.168.2.235 and 192.168.2.239 are clients and 192.168.2.52  on which
RTPproxy and Kamailio are running.

Awaiting your reply.

regards,
Ravi
RTP_Packet_loss.png
  




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