Re: [SR-Users] [EXT] Re: Using a network STUN Server

[Wilkins, Steve]

Thank you, I just was not sure what else would cause the relayed packets to not 
be sent out to my fios router.  As mentioned, I can pick any other server in my 
network and I can see, in the pcap file, that the relay is attempted to the 
selected server.  I verified our ACL and it is it open for TCP output to any IP 
and Port so that is not the issue.


-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Wednesday, December 26, 2018 12:55 PM
To: Kamailio (SER) - Users Mailing List 
Subject: [EXT] Re: [SR-Users] Using a network STUN Server

As STUN is a client-side construct, I am not aware of any functionality 
Kamailio might have to make *use* of a STUN server for its own representations 
about its reachability.

But I could be wrong. 

--
Sent from mobile. Apologies for brevity and errors. 

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[SR-Users] Using a network STUN Server

[Wilkins, Steve]

Hello All,

Can I point Kamilio to use a STUN server located on my network?  Kamilio will 
not relay to my fios router but will relay to other Servers on my network.

Thank you,
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[SR-Users] question about replace_body_all

[Wilkins, Steve]

Hello,

I am attempting to change the Message Body of an SIP message using 
replace_body_all;  I want to replace all attributes that start with  rtcp.
Here is my code =>

replace_body_all("rtcp:{1}[0-9]{5,}", newstring), and I call 
msg_apply_changes() after this call. Note that I verified that the regex is 
working using search.

The problem is that nothing gets changed. However, what I do get is a new 
string at the bottom of the Message Body with my newstring concatenated three 
times, which happens to be the number of times that rtcp occurs in the Message 
Body.

Note, I am doing this so that I can add the IP address to the rtcp attribute 
because this is required by a Provider.

Any ideas?

Thank you,
-Steve
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Re: [SR-Users] Double Session Description Protocol Version (v) 0 data when using rtpengine

[Wilkins, Steve]

Thank for your insight and honesty Alex!  I do appreciate feedback as it helps 
us grow...no matter our age.

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Wednesday, October 24, 2018 9:01 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

More generally, Steve, responding to some themes I see in your posts over the 
past few months:

You have a habit of posting overblown speculations about problems and painting 
yourself into grossly overcomplicated interpretations that send you down 
spurious paths of troubleshooting.

I think you need to seriously ponder Occam's Razor:

https://en.wikipedia.org/wiki/Occam%27s_razor

If you're running into an unexpected phenomenon, why do you assume it's a bug 
in Kamailio or a "de novo" problem? In fact, a little Googling will reveal that 
double SDP stanzas are a common question.

https://lists.kamailio.org/pipermail/sr-users/2015-August/089538.html

The simplest explanation is almost ALWAYS the correct one. Yes, occasionally it 
is possible to find a true, honest-to-god original bug in the product -- we all 
have. But it is not the explanation you should start with, nor one you should 
readily leap to, because it requires a lot more assumptions, so 
probabilistically, it is almost always the wrong theory. 

Consider it in terms of thermodynamics, mathematics, and the complexity 
introduced by additional moving parts if you like:

If you see an unexplained flash of light, there are numerous possible 
explanations. Some are quite simple: optical-neurological phenomenon, 
reflection from the mirror of a passing car, etc. Some are extremely 
complicated and involve novel, unproven ideas in physics or cosmology.

"Hey, what was that flash of ligth?"

Which should you start with FIRST? Think about it critically ...

-- Alex

On Wed, Oct 24, 2018 at 08:51:12AM -0400, Alex Balashov wrote:

> I think you've crawled down that rabbit hole about two millennia 
> prematurely. There is a much simpler explanation for why you're 
> getting a duplicated SDP stanza - in the logic of your route script. 
> It's not a bug in Kamailio.
> 
> I can't tell you exactly what the cause is, but I believe this avenue 
> of exploration will prove fruitful. It's a fairly common problem.
> 
> On Wed, Oct 24, 2018 at 12:49:55PM +, Wilkins, Steve wrote:
> 
> > I was looking at the sdpops_mod.c code hoping that there was an easy way to 
> > remove a sess_version (Session Description Protocol Version (v): 0) 
> > structure, but there does not appear to be that functionality in that 
> > module.  It might be a bit of work to remove a duplicate sess_version line.
> > 
> > I am using Kamailio 5.1, does anyone know if this is an issue in later 
> > versions?
> > 
> > Thank you,
> > -Steve
> 
> > ___
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> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> 
> 
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
> 
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> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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

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Re: [SR-Users] Double Session Description Protocol Version (v) 0 data when using rtpengine

[Wilkins, Steve]

Could you tell me what type of calls could cause this in a route script.  I 
just do see what I could be doing to generate that.  It only happens with 
rtpengine is enabled.

Thank you.

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Wednesday, October 24, 2018 8:51 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

I think you've crawled down that rabbit hole about two millennia prematurely. 
There is a much simpler explanation for why you're getting a duplicated SDP 
stanza - in the logic of your route script. It's not a bug in Kamailio. 

I can't tell you exactly what the cause is, but I believe this avenue of 
exploration will prove fruitful. It's a fairly common problem.

On Wed, Oct 24, 2018 at 12:49:55PM +, Wilkins, Steve wrote:

> I was looking at the sdpops_mod.c code hoping that there was an easy way to 
> remove a sess_version (Session Description Protocol Version (v): 0) 
> structure, but there does not appear to be that functionality in that module. 
>  It might be a bit of work to remove a duplicate sess_version line.
> 
> I am using Kamailio 5.1, does anyone know if this is an issue in later 
> versions?
> 
> Thank you,
> -Steve

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

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[SR-Users] Double Session Description Protocol Version (v) 0 data when using rtpengine

[Wilkins, Steve]

I was looking at the sdpops_mod.c code hoping that there was an easy way to 
remove a sess_version (Session Description Protocol Version (v): 0) structure, 
but there does not appear to be that functionality in that module.  It might be 
a bit of work to remove a duplicate sess_version line.

I am using Kamailio 5.1, does anyone know if this is an issue in later versions?

Thank you,
-Steve
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Re: [SR-Users] Double Session Description Protocol Version (v) 0 data when using rtpengine

[Wilkins, Steve]

Alex,

If this problem is a Kamailio "bug", is there a proper site to report it to?

Thank you

-Original Message-
From: sr-users  On Behalf Of Wilkins, Steve
Sent: Wednesday, October 24, 2018 7:44 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

Hi Alex,

I just ran a test, and yes, there is an error in the rtpengine log "Error 
Parsing RTP Header" and there is a double SDP present.

Thank you,
-Steve

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Wednesday, October 24, 2018 7:28 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

I would repeat my invitation to analyse RTPEngine logs for signs of a double 
offer/answer.

On Wed, Oct 24, 2018 at 11:03:11AM +, Wilkins, Steve wrote:

> I should also note that when the 200 OK is received from Asterisk, it does 
> not have the double SDP, only after Kamailio forwards the 200 OK does the 
> double appear.
> 
> -Steve
> 
> -Original Message-
> From: sr-users  On Behalf Of 
> Wilkins, Steve
> Sent: Tuesday, October 23, 2018 10:00 PM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] Double Session Description Protocol Version
> (v) 0 data when using rtpengine
> 
> No calls to fix_nated_sdp().
> 
> -Original Message-
> From: sr-users  On Behalf Of Alex 
> Balashov
> Sent: Tuesday, October 23, 2018 9:58 PM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] Double Session Description Protocol Version
> (v) 0 data when using rtpengine
> 
> Also, is there any possibility that you are calling fix_nated_sdp() and 
> rtpengine_offer/manage() successively? 
> 
> --
> Sent from mobile. Apologies for brevity and errors. 
> 
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Double Session Description Protocol Version (v) 0 data when using rtpengine

[Wilkins, Steve]

Hi Alex,

I just ran a test, and yes, there is an error in the rtpengine log "Error 
Parsing RTP Header" and there is a double SDP present.

Thank you,
-Steve

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Wednesday, October 24, 2018 7:28 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

I would repeat my invitation to analyse RTPEngine logs for signs of a double 
offer/answer.

On Wed, Oct 24, 2018 at 11:03:11AM +0000, Wilkins, Steve wrote:

> I should also note that when the 200 OK is received from Asterisk, it does 
> not have the double SDP, only after Kamailio forwards the 200 OK does the 
> double appear.
> 
> -Steve
> 
> -Original Message-
> From: sr-users  On Behalf Of 
> Wilkins, Steve
> Sent: Tuesday, October 23, 2018 10:00 PM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] Double Session Description Protocol Version 
> (v) 0 data when using rtpengine
> 
> No calls to fix_nated_sdp().
> 
> -Original Message-
> From: sr-users  On Behalf Of Alex 
> Balashov
> Sent: Tuesday, October 23, 2018 9:58 PM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] Double Session Description Protocol Version 
> (v) 0 data when using rtpengine
> 
> Also, is there any possibility that you are calling fix_nated_sdp() and 
> rtpengine_offer/manage() successively? 
> 
> --
> Sent from mobile. Apologies for brevity and errors. 
> 
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Double Session Description Protocol Version (v) 0 data when using rtpengine

[Wilkins, Steve]

I should also note that when the 200 OK is received from Asterisk, it does not 
have the double SDP, only after Kamailio forwards the 200 OK does the double 
appear.

-Steve

-Original Message-
From: sr-users  On Behalf Of Wilkins, Steve
Sent: Tuesday, October 23, 2018 10:00 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

No calls to fix_nated_sdp().

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Tuesday, October 23, 2018 9:58 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

Also, is there any possibility that you are calling fix_nated_sdp() and 
rtpengine_offer/manage() successively? 

--
Sent from mobile. Apologies for brevity and errors. 

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Re: [SR-Users] Double Session Description Protocol Version (v) 0 data when using rtpengine

[Wilkins, Steve]

No calls to fix_nated_sdp().

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Tuesday, October 23, 2018 9:58 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

Also, is there any possibility that you are calling fix_nated_sdp() and 
rtpengine_offer/manage() successively? 

--
Sent from mobile. Apologies for brevity and errors. 

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Re: [SR-Users] Double Session Description Protocol Version (v) 0 data when using rtpengine

[Wilkins, Steve]

Let me check...Thank you.

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Tuesday, October 23, 2018 9:58 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

Also, is there any possibility that you are calling fix_nated_sdp() and 
rtpengine_offer/manage() successively? 

--
Sent from mobile. Apologies for brevity and errors. 

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Re: [SR-Users] Double Session Description Protocol Version (v) 0 data when using rtpengine

[Wilkins, Steve]

I thought using data would imply what I meant to say, but I guess I was wrong.  
Sometime I take to many shortcuts in typing.  

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Tuesday, October 23, 2018 9:57 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

Then your description of the problem, and the subject line of your post, is not 
accurate. :-) 

This should not be happening. What does the rtpengine log show for the given 
Call-ID? 

--
Sent from mobile. Apologies for brevity and errors. 

-Original Message-
From: "Wilkins, Steve" 
To: "Kamailio (SER) - Users Mailing List" 
Sent: Tue, 23 Oct 2018 9:55 PM
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

Entire section.

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Tuesday, October 23, 2018 9:31 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

Is the entire SDP body doubled, or just the v=0 line? 

--
Sent from mobile. Apologies for brevity and errors. 

-Original Message-----
From: "Wilkins, Steve" 
To: "Kamailio (SER) - Users Mailing List" 
Sent: Tue, 23 Oct 2018 9:07 PM
Subject: [SR-Users] Double Session Description Protocol Version (v) 0 data when 
using rtpengine

Hello all,

I noticed double Session Description Protocol Version (v) 0 data in the SDP 
section when using rtpengine with Kamailio.  Has any else noticed this?  Is 
there a way for Kamailio to remove one of them?

Thank you,
-Steve

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Re: [SR-Users] Double Session Description Protocol Version (v) 0 data when using rtpengine

[Wilkins, Steve]

Entire section.

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Tuesday, October 23, 2018 9:31 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data 
when using rtpengine

Is the entire SDP body doubled, or just the v=0 line? 

--
Sent from mobile. Apologies for brevity and errors. 

-Original Message-
From: "Wilkins, Steve" 
To: "Kamailio (SER) - Users Mailing List" 
Sent: Tue, 23 Oct 2018 9:07 PM
Subject: [SR-Users] Double Session Description Protocol Version (v) 0 data when 
using rtpengine

Hello all,

I noticed double Session Description Protocol Version (v) 0 data in the SDP 
section when using rtpengine with Kamailio.  Has any else noticed this?  Is 
there a way for Kamailio to remove one of them?

Thank you,
-Steve

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Re: [SR-Users] Kamailio forwarding call via Public IP Address vs Private IP Address

[Wilkins, Steve]

Please ignore this question...error on my part.  Sorry!

From: sr-users  On Behalf Of Wilkins, Steve
Sent: Sunday, October 21, 2018 9:30 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Kamailio forwarding call via Public IP Address vs 
Private IP Address

Sorry I meant forwarded to Kamailio Public IP Address, not Asterisk Public IP

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: Sunday, October 21, 2018 7:16 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Kamailio forwarding call via Public IP Address vs Private 
IP Address

Good  morning All,

I have two almost Identical (Kamailio=>Asterisk) systems set up, but one of 
them is receiving calls and forwarding them to the Asterisk Public IP, and the 
other is Forwarding to the Asterisk Private IP.  The call is coming in to the 
number and domain in both scenarios.  Any ideas on what would cause this?

Thank you,
-Steve
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[SR-Users] Kamailio forwarding call via Public IP Address vs Private IP Address

[Wilkins, Steve]

Good  morning All,

I have two almost Identical (Kamailio=>Asterisk) systems set up, but one of 
them is receiving calls and forwarding them to the Asterisk Public IP, and the 
other is Forwarding to the Asterisk Private IP.  The call is coming in to the 
number and domain in both scenarios.  Any ideas on what would cause this?

Thank you,
-Steve
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[SR-Users] db_mysql when compiling Kamailio

[Wilkins, Steve]

Hello All,

When  db_mysql is selected in the make, the make looks for -lmariadb. However, 
I want MySQL, not mariadb.  Is there a way to let the make know that MySQL is 
preferred ov mariadb?

Thank you
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[SR-Users] RTPEngine Question

[Wilkins, Steve]

Hello All,

I have an issue where Kamailio-RTPEngine-Asterisk calls work good when a 
softphone UAC on an IOS phone makes an Inbound call to a WebRTC client.  
However, if the softphone UAC is on Windows, it does not work (No Audio/Video). 
 I noticed in the Wireshark trace that when using IOS I see STUN packets, but 
when using Windows I see no STUN packets.  I believe this may be my issue, but 
I am not sure why the IOS would create STUN packets but when using Windows I 
get no STUN packets.  Does anyone have any thoughts on what could be causing 
this?

Thank you,

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Re: [SR-Users] re- double record route

[Wilkins, Steve]

Hi Alex,  

I did try removing the records routes, but I still received the '404 Not Here' 
response from the Server; I thought maybe it was an issue with the transport 
type, but I tried both UDP and TCP...with no luck.  The BYE is definitely being 
sent to the correct Server and Port, this is why I thought it had some weird 
thing to do with the Record Route.

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Wednesday, September 19, 2018 10:29 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] re- double record route

Record-Routes are present only in dialog-forming - that is to say, initial - 
INVITEs and their replies. A BYE is an in-dialog request and should have a 
Route set constructed from those RRs, but should not contain any RRs.

On Wed, Sep 19, 2018 at 02:26:27PM +, Wilkins, Steve wrote:

> Hi Mojtaba,
> 
> I will try to explain the situation a little better.
> 
> A softphone sends an 'INVITE' to a WebRTC Client, the call comes to Kamailio, 
> which then forwards it to Asterisk, and then Asterisk to the WebRTC Client.  
> The softphone is connected to a Providers Server (IP 10.20.5.5) and it sends 
> the Record Routes of (10.20.5.5;lr=on;r2=on), and 
> (10.20.5.5;transport=tcp;lr=on;r2=on) in the initial 'INVITE'.  The problem 
> is, if the WebRTC client initiates the 'BYE', when Kamilio forwards the 
> 'BYE', the Records Route's form the initial 'INVITE' are missing and the 
> Provider ( softphone) returns "404 Not Here".
> 
> Thank you,
> -Steve
> 
> -Original Message-
> From: sr-users  On Behalf Of 
> Mojtaba
> Sent: Monday, September 17, 2018 2:51 AM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] re- double record route
> 
> Your response confused me, But let me describe some rules.
> The BYE request is used the record route header in it's INVITE request in 
> Dialogs. Then when UE want to send BYE, It add a all record route in BYE 
> message.
> On other hand, the SIP stack that UE is used, may be different, So you must 
> be sure about it. You have two choice: Diving to SIP stack in UE (doesn't 
> good choice), and use the power of Kamailio.
> For example, Use Path header to save the second record route (or your 
> favourite record route) in INVITE message, then if the UE send BYE message 
> which the Path header in, use this Path header for relaying in kamailio.
> As i said, It is needed to paste the log for better understanding.
> With Regards.Mojtaba
> On Mon, Sep 17, 2018 at 2:17 AM Wilkins, Steve  wrote:
> >
> > Hi Mojtaba,
> >
> > But when I send the 'BYE' doesn't the double Record-Route from the 'INVITE' 
> > (from IOS) need to be there, so that IOS can find it's Proxies and complete 
> > the transaction and send back a '200 OK'?
> >
> > Thank you,
> >
> > -Original Message-
> > From: sr-users  On Behalf Of 
> > Mojtaba
> > Sent: Sunday, September 16, 2018 12:47 PM
> > To: Kamailio (SER) - Users Mailing List 
> > 
> > Subject: Re: [SR-Users] re- double record route
> >
> > What do you mean exactly? Do you mean the INVITE is received from IOS has 
> > double record route? If this true, the Kamailio remove all record route 
> > when the INVITE request is received by default. In other words, Kamailio 
> > remove all record routes form downstream or upstream in INVITE request by 
> > default.
> > You should paste a simple wireshark to solve it as soon.
> > With Regards.Mojtaba
> > On Sun, Sep 16, 2018 at 6:10 PM Wilkins, Steve  wrote:
> > >
> > > Hi Henning,
> > >
> > >
> > >
> > > Yes I do have that enabled.  What is happening is that one of the 
> > > providers on IOS is sending a double record route on the INVITE, but it 
> > > is getting lost somewhere so when I send a 'BYE', I get a "404 not here". 
> > >  When I look at the SIP message I see only one of the record routes from 
> > > the INVITE, therefore my assumption is, this is what is causing the 404, 
> > > because everything else looks good.
> > >
> > >
> > >
> > > ___
> > > Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org 
> > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> >
> >
> >
> > --
> > --Mojtaba Esfandiari.S
> >
> > ___
> > Kamailio (SER) - Users Mailing List
> > sr-users@lists.kamailio.org
> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> > ___
> > Kamaili

Re: [SR-Users] re- double record route

[Wilkins, Steve]

Hi Mojtaba,

I will try to explain the situation a little better.

A softphone sends an 'INVITE' to a WebRTC Client, the call comes to Kamailio, 
which then forwards it to Asterisk, and then Asterisk to the WebRTC Client.  
The softphone is connected to a Providers Server (IP 10.20.5.5) and it sends 
the Record Routes of (10.20.5.5;lr=on;r2=on), and 
(10.20.5.5;transport=tcp;lr=on;r2=on) in the initial 'INVITE'.  The problem is, 
if the WebRTC client initiates the 'BYE', when Kamilio forwards the 'BYE', the 
Records Route's form the initial 'INVITE' are missing and the Provider ( 
softphone) returns "404 Not Here".

Thank you,
-Steve

-Original Message-
From: sr-users  On Behalf Of Mojtaba
Sent: Monday, September 17, 2018 2:51 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] re- double record route

Your response confused me, But let me describe some rules.
The BYE request is used the record route header in it's INVITE request in 
Dialogs. Then when UE want to send BYE, It add a all record route in BYE 
message.
On other hand, the SIP stack that UE is used, may be different, So you must be 
sure about it. You have two choice: Diving to SIP stack in UE (doesn't good 
choice), and use the power of Kamailio.
For example, Use Path header to save the second record route (or your favourite 
record route) in INVITE message, then if the UE send BYE message which the Path 
header in, use this Path header for relaying in kamailio.
As i said, It is needed to paste the log for better understanding.
With Regards.Mojtaba
On Mon, Sep 17, 2018 at 2:17 AM Wilkins, Steve  wrote:
>
> Hi Mojtaba,
>
> But when I send the 'BYE' doesn't the double Record-Route from the 'INVITE' 
> (from IOS) need to be there, so that IOS can find it's Proxies and complete 
> the transaction and send back a '200 OK'?
>
> Thank you,
>
> -Original Message-
> From: sr-users  On Behalf Of 
> Mojtaba
> Sent: Sunday, September 16, 2018 12:47 PM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] re- double record route
>
> What do you mean exactly? Do you mean the INVITE is received from IOS has 
> double record route? If this true, the Kamailio remove all record route when 
> the INVITE request is received by default. In other words, Kamailio remove 
> all record routes form downstream or upstream in INVITE request by default.
> You should paste a simple wireshark to solve it as soon.
> With Regards.Mojtaba
> On Sun, Sep 16, 2018 at 6:10 PM Wilkins, Steve  wrote:
> >
> > Hi Henning,
> >
> >
> >
> > Yes I do have that enabled.  What is happening is that one of the providers 
> > on IOS is sending a double record route on the INVITE, but it is getting 
> > lost somewhere so when I send a 'BYE', I get a "404 not here".  When I look 
> > at the SIP message I see only one of the record routes from the INVITE, 
> > therefore my assumption is, this is what is causing the 404, because 
> > everything else looks good.
> >
> >
> >
> > ___
> > Kamailio (SER) - Users Mailing List
> > sr-users@lists.kamailio.org
> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> --Mojtaba Esfandiari.S
>
> ___
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> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> ___
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> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users



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[SR-Users] double record route from INVITE

[Wilkins, Steve]

Hi Mojtaba,



But when I send the 'BYE' doesn't the double Record-Route from the 'INVITE' 
(from IOS) need to be there, so that IOS can find it's Proxies and complete the 
transaction and send back a '200 OK'?



Thank you,

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Re: [SR-Users] re- double record route

[Wilkins, Steve]

Hi Mojtaba,

But when I send the 'BYE' doesn't the double Record-Route from the 'INVITE' 
(from IOS) need to be there, so that IOS can find it's Proxies and complete the 
transaction and send back a '200 OK'?

Thank you, 

-Original Message-
From: sr-users  On Behalf Of Mojtaba
Sent: Sunday, September 16, 2018 12:47 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] re- double record route

What do you mean exactly? Do you mean the INVITE is received from IOS has 
double record route? If this true, the Kamailio remove all record route when 
the INVITE request is received by default. In other words, Kamailio remove all 
record routes form downstream or upstream in INVITE request by default.
You should paste a simple wireshark to solve it as soon.
With Regards.Mojtaba
On Sun, Sep 16, 2018 at 6:10 PM Wilkins, Steve  wrote:
>
> Hi Henning,
>
>
>
> Yes I do have that enabled.  What is happening is that one of the providers 
> on IOS is sending a double record route on the INVITE, but it is getting lost 
> somewhere so when I send a 'BYE', I get a "404 not here".  When I look at the 
> SIP message I see only one of the record routes from the INVITE, therefore my 
> assumption is, this is what is causing the 404, because everything else looks 
> good.
>
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users



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Re: [SR-Users] re- double record route

[Wilkins, Steve]

The routes from the 'INVITE' were changed by something.  The 'BYE' route does 
not have the routes from the 'INVITE', I don’t know how they got deleted.

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Sunday, September 16, 2018 10:05 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] re- double record route

Route sets need to be fastidiously and scrupulously conserved by both UAs on 
both sides in all in-dialog requests.

On Sun, Sep 16, 2018 at 01:40:24PM +, Wilkins, Steve wrote:

> Hi Henning,
> 
> 
> 
> Yes I do have that enabled.  What is happening is that one of the providers 
> on IOS is sending a double record route on the INVITE, but it is getting lost 
> somewhere so when I send a 'BYE', I get a "404 not here".  When I look at the 
> SIP message I see only one of the record routes from the INVITE, therefore my 
> assumption is, this is what is causing the 404, because everything else looks 
> good.
> 

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

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[SR-Users] re- double record route

[Wilkins, Steve]

Hi Henning,



Yes I do have that enabled.  What is happening is that one of the providers on 
IOS is sending a double record route on the INVITE, but it is getting lost 
somewhere so when I send a 'BYE', I get a "404 not here".  When I look at the 
SIP message I see only one of the record routes from the INVITE, therefore my 
assumption is, this is what is causing the 404, because everything else looks 
good.

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[SR-Users] double record route?

[Wilkins, Steve]

Good Morning All,

Is there any way to add a double record route?  I tried adding a second record 
route and I always only get the first one added.
I have tried record_route(), and record_route_advertised_address(...), but 
I still only get the first record route added.

Thank you,
-Steve
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Re: [SR-Users] Record Routes

[Wilkins, Steve]

Alex, you are correct!

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Tuesday, September 11, 2018 8:49 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Record Routes

I think you are treading into dangerous territory that stems from a lack of 
understanding of SIP. You can't just go changing stuff to look like something 
else.

BYEs are in-dialog requests and their Route: sets are constructed from 
Record-Route sets set up during the initial INVITE transaction that formed the 
dialog. They need to be left alone.

On Tue, Sep 11, 2018 at 12:37:56PM +, Wilkins, Steve wrote:

> Thank you,
> 
> I am doing this just as a test, because I cannot get a soft-phone to hang up 
> when the 'BYE' is initiated by a WebRTC client (although, some provider 
> soft-phones do work ,Hang up that is).
> 
> I have pcap logs of the one that works, so I am trying to get the SIP 
> message to match that one.  The only difference between the one that works 
> and the one that does not, is that the Route's: are different.
> 
> I must be misunderstanding the SIP Conversation because, the 'BYE' request is 
> getting to the soft-phone provider, but the provider is sending back '404 Not 
> Here'.  My confusion is that since the Request is making it to the provider, 
> why does the Request Route matter at this point.
> 
> -Steve
> 
> -Original Message-
> From: sr-users  On Behalf Of Alex 
> Balashov
> Sent: Tuesday, September 11, 2018 7:25 AM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] Record Routes
> 
> On Tue, Sep 11, 2018 at 11:22:26AM +, Wilkins, Steve wrote:
> 
> > Can a Route: be removed?, if so, how?
> 
>if(is_present_hf("Route"))
>   remove_hf("Route");
> 
> Some may view the if-statement as redundant, given that remove_hf() will 
> simply have zero effect if the header is not present, without complaint.
> 
> But unless you are doing this in the context of an initial request, you need 
> to carefully ask yourself about your motivations.
> 
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
> 
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] kamailio.cfg errors with this example from documentation

[Wilkins, Steve]

Sorry Alex, you are correct, I spaced that out.

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Friday, September 7, 2018 10:57 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] kamailio.cfg errors with this example from documentation

On Fri, Sep 07, 2018 at 02:55:04PM +, Wilkins, Steve wrote:

> Did I miss something, what was your way 

Yes, you clearly did not read my responses. :-)

> $var(x) = $ct;
> $(var(x){s.substr,1,0});

   $var(x) = $ct;
   $dlg_var(callercontact) = $(var(x){s.substr,1,0});

Or in the interest of simplicity:

   $dlg_var(callercontact) = $(ct{s.substr,1,0});

-- 
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
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Re: [SR-Users] kamailio.cfg errors with this example from documentation

[Wilkins, Steve]

Did I miss something, what was your way 

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Friday, September 7, 2018 10:52 AM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] kamailio.cfg errors with this example from documentation

On Fri, Sep 07, 2018 at 02:47:22PM +, Wilkins, Steve wrote:
> Here is what I actually do =>
> 
> $var(x) = $ct;
> $(var(x){s.substr,1,0});
> $dlg_var(callercontact) = $var(x);

Well, you can't do that. You have to do it my way. ;-)

-- 
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
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Re: [SR-Users] kamailio.cfg errors with this example from documentation

[Wilkins, Steve]

Here is what I actually do =>

$var(x) = $ct;
$(var(x){s.substr,1,0});
$dlg_var(callercontact) = $var(x);

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Friday, September 7, 2018 10:41 AM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] kamailio.cfg errors with this example from documentation

If I understand you correctly, you're saying ...

   $dlg_var(x) = $(var(x){s.substr,1,0});

doesn't work?

Bear in mind you can't just copy and paste the example. This, from the example, 
will not work:

   $(var(x){s.substr,1,0});

... because Kamailio's grammar doesn't allow for it.

-- Alex

On Fri, Sep 07, 2018 at 02:36:50PM +0000, Wilkins, Steve wrote:

> Yes, I  will assign it to
> $dlg_var(X) = $var(x);
> 
> -Original Message-
> From: sr-users  On Behalf Of Alex 
> Balashov
> Sent: Friday, September 7, 2018 10:31 AM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] kamailio.cfg errors with this example from 
> documentation
> 
> On Fri, Sep 07, 2018 at 02:20:11PM +, Wilkins, Steve wrote:
> 
> > 0(1) CRITICAL:  [core/cfg.y:3489]: yyerror_at(): parse 
> > error in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, 
> > column
> > 1-23: pvar with transformations in assignment left side
> 
> This is the real issue. Are you doing anything with the transformed value, 
> like storing it in another variable? It cannot exist as a free-floating 
> statement.
> 
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
> 
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Re: [SR-Users] kamailio.cfg errors with this example from documentation

[Wilkins, Steve]

Yes, I  will assign it to 
$dlg_var(X) = $var(x);

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Friday, September 7, 2018 10:31 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] kamailio.cfg errors with this example from documentation

On Fri, Sep 07, 2018 at 02:20:11PM +, Wilkins, Steve wrote:

> 0(1) CRITICAL:  [core/cfg.y:3489]: yyerror_at(): parse error 
> in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column
> 1-23: pvar with transformations in assignment left side

This is the real issue. Are you doing anything with the transformed value, like 
storing it in another variable? It cannot exist as a free-floating statement.

--
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[SR-Users] kamailio.cfg errors with this example from documentation

[Wilkins, Steve]

$var(x) = "abcd";
$(var(x){s.substr,1,0});
Hello all,

I took the example from the documentation, but Kamailio.cfg has errors with 
this example.

Errors=>
0(1) CRITICAL:  [core/cfg.y:3489]: yyerror_at(): parse error in 
config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 1-23: pvar 
with transformations in assignment left side
0(1) CRITICAL:  [core/cfg.y:3492]: yyerror_at(): parse error in 
config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 24: syntax 
error
0(1) CRITICAL:  [core/cfg.y:3492]: yyerror_at(): parse error in 
config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 24: bad 
command

Thank you,

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Re: [SR-Users] handle_ruri_alias() question or issue?

[Wilkins, Steve]

Hi Joel,

Can I contact you at your email address?

Thanks,
-Steve

From: sr-users  On Behalf Of Joel Serrano
Sent: Sunday, September 2, 2018 12:26 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?

Hi Steve,

Can you send a pcap of the call so I can have a look at it? I would like to see 
what arrives and leaves kamailio to have an idea of what is going on..

I have the feeling something is not correct in the config, can you post the 
relevant parts or are you using default kamailio config?


On Sun, Sep 2, 2018 at 05:07 Wilkins, Steve 
mailto:swwilk...@mitre.org>> wrote:
$du: sip:20.20.20.20;transport=tcp;lr;r2=on;ftag=as761c30aa;nat=yes
$ru: sip:125.10.1.14;lr;r2=on

handle_ruri_alias(); has no effect
$mb =>
[BYE 
sip:2135601240@10.10.10.10:5060<http://sip:2135601240@10.10.10.10:5060>;alias=125.10.1.14~38869~2
 SIP/2.0#015#012
Via: SIP/2.0/TCP 
30.30.30.30:5060;rport;branch=z9hG4bKPjd336f9fd-d890-4017-8f38-c668fb067823;alias#015#012
From: 
mailto:sip%3A7032935282@125.10.1.14>>;tag=a09b41bb-f621-47e1-9753-3efc6e00268e#015#012
To: "Mitre6 Testing" 
mailto:sip%3A2135601240@125.10.1.14>>;tag=as761c30aa#015#012
Call-ID: 
00ef3468081a927b7512d9445badc0f8@125.10.1.14#015#012<http://00ef3468081a927b7512d9445badc0f8@125.10.1.14#015%23012>
CSeq: 30144 BYE#015#012Route: 
#015#012
Route: #015#012
Route: #015#012
Route: #015#012Reason: Q.850;cause=16#015#012
Max-Forwards: 69#015#012
User-Agent: Asterisk PBX 15.3.0#015#012Content-Length:  0

Kamailio Public IP: 20.20.20.20
Kamailio Private IP 10.10.10.10
Asterisk  Private IP:30.30.30.30
IP of Called Server: 125.10.1.14

Called Number: 2135601240

I want the relay to go to the alias: 125.10.1.14:38869<http://125.10.1.14:38869>

Thank you!

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Joel Serrano
Sent: Saturday, September 1, 2018 9:41 PM

To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?

Can you print $ru and $du of that BYE in the logs and send them here?



On Sat, Sep 1, 2018 at 14:41 Wilkins, Steve 
mailto:swwilk...@mitre.org>> wrote:
Right before t_relay, $mb =>

[BYE sip:3128145656@10.10.10.10:5060;alias=125.10.1.15~32940~2 
SIP/2.0#015#012Via: SIP/2.0/TCP 
172.21.1.124:5060;rport;branch=z9hG4bKPj88f9c57d-5db6-4731-83c9-df478782fa39;alias#015#012From:
 
mailto:sip%3A7032935282@125.10.1.15>>;tag=aed37e42-1af3-4944-9132-74ec323ceda3#015#012To:
 "Mitre6 Testing" 
mailto:sip%3A3128145656@125.10.1.15>>;tag=as09387c4a#015#012Call-ID:
 
6f6147b437dfae9c7b5731d1679189ed@125.10.1.15#015#012CSeq<http://6f6147b437dfae9c7b5731d1679189ed@125.10.1.15#015%23012CSeq>:
 27644 BYE#015#012Route: 
#015#012Route: 
#015#012Route: 
#015#012Route: 
#015#012Reason: Q.850;cause=16#015#012Max-Forwards: 
69#015#012User-Agent: Asterisk PBX 15.3.0#015#012Content-Length:  0

I want the BYE to be sent to the alias 
125.10.1.15:32940<http://125.10.1.15:32940>

Nothing I do seems to let me get that alias and send the BYE to that 
address:port

Thank you,
-Steve

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Igor Olhovskiy
Sent: Saturday, September 1, 2018 2:05 PM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?

What is in $du if you log it right after handle_ruri_alias()?
Maybe you overwrite it somewhere later?

Regards, Igor
On Sep 1, 2018, 7:34 PM +0200, Wilkins, Steve 
mailto:swwilk...@mitre.org>>, wrote:
Thank you Joel,

My issue is that, given the Incoming Request: [[BYE 
sip:3128145656@20.20.20.20:5060<http://sip:2406506175@20.20.20.20:5060>;alias=10.10.10.5~55157~2
 SIP/2.0.  I want to route the BYE to the alias 
10.10.10.5:55157<http://10.10.10.5:55157>.  I have tried the combination you 
spoke of and it always still get routed to 20.20.20.20.5060.

I have even tried storing the original Via from the INVITE and using that in 
the BYE, however, I get a variable to maintain it’s state.

Thanks again,
-Steve

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Joel Serrano
Sent: Saturday, September 1, 2018 10:43 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?

I think you are mixing 2 different approaches of achieving the same:

1- Use: fix_nated_contact() --> Will *modify* the contact header and *replace* 
current ip:port with correct ip:port.

2- Use: set_contact_alias() in conjunction with handle_ruri_alias() --> 
set_contact_alias will *add an alias* with correct ip:port but will not 
*modify* the current ip:port sent by the UAC.  handle_ruri_alias will search 
for an alias in the RURI and if found, it will remove it and *s

Re: [SR-Users] handle_ruri_alias() question or issue?

[Wilkins, Steve]

$du: sip:20.20.20.20;transport=tcp;lr;r2=on;ftag=as761c30aa;nat=yes
$ru: sip:125.10.1.14;lr;r2=on

handle_ruri_alias(); has no effect
$mb =>
[BYE sip:2135601240@10.10.10.10:5060;alias=125.10.1.14~38869~2 SIP/2.0#015#012
Via: SIP/2.0/TCP 
30.30.30.30:5060;rport;branch=z9hG4bKPjd336f9fd-d890-4017-8f38-c668fb067823;alias#015#012
From: 
;tag=a09b41bb-f621-47e1-9753-3efc6e00268e#015#012
To: "Mitre6 Testing" ;tag=as761c30aa#015#012
Call-ID: 00ef3468081a927b7512d9445badc0f8@125.10.1.14#015#012
CSeq: 30144 BYE#015#012Route: 
#015#012
Route: #015#012
Route: #015#012
Route: #015#012Reason: Q.850;cause=16#015#012
Max-Forwards: 69#015#012
User-Agent: Asterisk PBX 15.3.0#015#012Content-Length:  0

Kamailio Public IP: 20.20.20.20
Kamailio Private IP 10.10.10.10
Asterisk  Private IP:30.30.30.30
IP of Called Server: 125.10.1.14

Called Number: 2135601240

I want the relay to go to the alias: 125.10.1.14:38869

Thank you!

From: sr-users  On Behalf Of Joel Serrano
Sent: Saturday, September 1, 2018 9:41 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?

Can you print $ru and $du of that BYE in the logs and send them here?



On Sat, Sep 1, 2018 at 14:41 Wilkins, Steve 
mailto:swwilk...@mitre.org>> wrote:
Right before t_relay, $mb =>

[BYE sip:3128145656@10.10.10.10:5060;alias=125.10.1.15~32940~2 
SIP/2.0#015#012Via: SIP/2.0/TCP 
172.21.1.124:5060;rport;branch=z9hG4bKPj88f9c57d-5db6-4731-83c9-df478782fa39;alias#015#012From:
 
mailto:sip%3A7032935282@125.10.1.15>>;tag=aed37e42-1af3-4944-9132-74ec323ceda3#015#012To:
 "Mitre6 Testing" 
mailto:sip%3A3128145656@125.10.1.15>>;tag=as09387c4a#015#012Call-ID:
 
6f6147b437dfae9c7b5731d1679189ed@125.10.1.15#015#012CSeq<http://6f6147b437dfae9c7b5731d1679189ed@125.10.1.15#015%23012CSeq>:
 27644 BYE#015#012Route: 
#015#012Route: 
#015#012Route: 
#015#012Route: 
#015#012Reason: Q.850;cause=16#015#012Max-Forwards: 
69#015#012User-Agent: Asterisk PBX 15.3.0#015#012Content-Length:  0

I want the BYE to be sent to the alias 
125.10.1.15:32940<http://125.10.1.15:32940>

Nothing I do seems to let me get that alias and send the BYE to that 
address:port

Thank you,
-Steve

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Igor Olhovskiy
Sent: Saturday, September 1, 2018 2:05 PM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?

What is in $du if you log it right after handle_ruri_alias()?
Maybe you overwrite it somewhere later?

Regards, Igor
On Sep 1, 2018, 7:34 PM +0200, Wilkins, Steve 
mailto:swwilk...@mitre.org>>, wrote:
Thank you Joel,

My issue is that, given the Incoming Request: [[BYE 
sip:3128145656@20.20.20.20:5060<http://sip:2406506175@20.20.20.20:5060>;alias=10.10.10.5~55157~2
 SIP/2.0.  I want to route the BYE to the alias 
10.10.10.5:55157<http://10.10.10.5:55157>.  I have tried the combination you 
spoke of and it always still get routed to 20.20.20.20.5060.

I have even tried storing the original Via from the INVITE and using that in 
the BYE, however, I get a variable to maintain it’s state.

Thanks again,
-Steve

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Joel Serrano
Sent: Saturday, September 1, 2018 10:43 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?

I think you are mixing 2 different approaches of achieving the same:

1- Use: fix_nated_contact() --> Will *modify* the contact header and *replace* 
current ip:port with correct ip:port.

2- Use: set_contact_alias() in conjunction with handle_ruri_alias() --> 
set_contact_alias will *add an alias* with correct ip:port but will not 
*modify* the current ip:port sent by the UAC.  handle_ruri_alias will search 
for an alias in the RURI and if found, it will remove it and *set $du 
accordingly*.You will not see the contact with correct ip:port in RURI but 
routing will be sent to correct ip:port ($du).

This is a high level explanation, but I hope it helps you understand what is 
going on. If you print $du in the logs after calling handle_ruri_alias() you 
should see what you expect (if you previously called set_contact_alias)... Or 
you can call set_contact_alias on not call handle_ruri_alias. Two ways of doing 
the same.

On Sat, Sep 1, 2018 at 04:47 Wilkins, Steve 
mailto:swwilk...@mitre.org>> wrote:
Good Morning All,

The following Incoming request came in =>
SIP Incoming Request: [[BYE 
sip:3128145656@20.20.20.20:5060<http://sip:2406506175@20.20.20.20:5060>;alias=10.10.10.5~55157~2
 SIP/2.0

I called handle_ruri_alias(), and expected the destination and port to be set 
to 10.10.10.5:55157<http://10.10.10.5:55157>, but it was not.
It was left 20.20.20.20:5060<http://20.20.

Re: [SR-Users] handle_ruri_alias() question or issue?

[Wilkins, Steve]

Right before t_relay, $mb =>

[BYE sip:3128145656@10.10.10.10:5060;alias=125.10.1.15~32940~2 
SIP/2.0#015#012Via: SIP/2.0/TCP 
172.21.1.124:5060;rport;branch=z9hG4bKPj88f9c57d-5db6-4731-83c9-df478782fa39;alias#015#012From:
 
;tag=aed37e42-1af3-4944-9132-74ec323ceda3#015#012To:
 "Mitre6 Testing" ;tag=as09387c4a#015#012Call-ID: 
6f6147b437dfae9c7b5731d1679189ed@125.10.1.15#015#012CSeq: 27644 
BYE#015#012Route: 
#015#012Route: 
#015#012Route: 
#015#012Route: 
#015#012Reason: Q.850;cause=16#015#012Max-Forwards: 
69#015#012User-Agent: Asterisk PBX 15.3.0#015#012Content-Length:  0

I want the BYE to be sent to the alias 125.10.1.15:32940

Nothing I do seems to let me get that alias and send the BYE to that 
address:port

Thank you,
-Steve

From: sr-users  On Behalf Of Igor Olhovskiy
Sent: Saturday, September 1, 2018 2:05 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?

What is in $du if you log it right after handle_ruri_alias()?
Maybe you overwrite it somewhere later?

Regards, Igor
On Sep 1, 2018, 7:34 PM +0200, Wilkins, Steve 
mailto:swwilk...@mitre.org>>, wrote:

Thank you Joel,

My issue is that, given the Incoming Request: [[BYE 
sip:3128145656@20.20.20.20:5060<http://sip:2406506175@20.20.20.20:5060>;alias=10.10.10.5~55157~2
 SIP/2.0.  I want to route the BYE to the alias 10.10.10.5:55157.  I have tried 
the combination you spoke of and it always still get routed to 20.20.20.20.5060.

I have even tried storing the original Via from the INVITE and using that in 
the BYE, however, I get a variable to maintain it’s state.

Thanks again,
-Steve

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Joel Serrano
Sent: Saturday, September 1, 2018 10:43 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?

I think you are mixing 2 different approaches of achieving the same:

1- Use: fix_nated_contact() --> Will *modify* the contact header and *replace* 
current ip:port with correct ip:port.

2- Use: set_contact_alias() in conjunction with handle_ruri_alias() --> 
set_contact_alias will *add an alias* with correct ip:port but will not 
*modify* the current ip:port sent by the UAC.  handle_ruri_alias will search 
for an alias in the RURI and if found, it will remove it and *set $du 
accordingly*.You will not see the contact with correct ip:port in RURI but 
routing will be sent to correct ip:port ($du).

This is a high level explanation, but I hope it helps you understand what is 
going on. If you print $du in the logs after calling handle_ruri_alias() you 
should see what you expect (if you previously called set_contact_alias)... Or 
you can call set_contact_alias on not call handle_ruri_alias. Two ways of doing 
the same.

On Sat, Sep 1, 2018 at 04:47 Wilkins, Steve 
mailto:swwilk...@mitre.org>> wrote:
Good Morning All,

The following Incoming request came in =>
SIP Incoming Request: [[BYE 
sip:3128145656@20.20.20.20:5060<http://sip:2406506175@20.20.20.20:5060>;alias=10.10.10.5~55157~2
 SIP/2.0

I called handle_ruri_alias(), and expected the destination and port to be set 
to 10.10.10.5:55157<http://10.10.10.5:55157>, but it was not.
It was left 20.20.20.20:5060<http://20.20.20.20:5060>.

Is my thinking correct or is there another way to set the destination and port 
to the alias?

Thanks All!,
-Steve
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Re: [SR-Users] handle_ruri_alias() question or issue?

[Wilkins, Steve]

Thank you Joel,

My issue is that, given the Incoming Request: [[BYE 
sip:2406506175@20.20.20.20:5060<http://sip:2406506175@20.20.20.20:5060>;alias=10.10.10.5~55157~2
 SIP/2.0.  I want to route the BYE to the alias 10.10.10.5:55157.  I have tried 
the combination you spoke of and it always still get routed to 20.20.20.20.5060.

I have even tried storing the original Via from the INVITE and using that in 
the BYE, however, I get a variable to maintain it’s state.

Thanks again,
-Steve

From: sr-users  On Behalf Of Joel Serrano
Sent: Saturday, September 1, 2018 10:43 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?

I think you are mixing 2 different approaches of achieving the same:

1- Use: fix_nated_contact() --> Will *modify* the contact header and *replace* 
current ip:port with correct ip:port.

2- Use: set_contact_alias() in conjunction with handle_ruri_alias() --> 
set_contact_alias will *add an alias* with correct ip:port but will not 
*modify* the current ip:port sent by the UAC.  handle_ruri_alias will search 
for an alias in the RURI and if found, it will remove it and *set $du 
accordingly*.You will not see the contact with correct ip:port in RURI but 
routing will be sent to correct ip:port ($du).

This is a high level explanation, but I hope it helps you understand what is 
going on. If you print $du in the logs after calling handle_ruri_alias() you 
should see what you expect (if you previously called set_contact_alias)... Or 
you can call set_contact_alias on not call handle_ruri_alias. Two ways of doing 
the same.

On Sat, Sep 1, 2018 at 04:47 Wilkins, Steve 
mailto:swwilk...@mitre.org>> wrote:
Good Morning All,

The following Incoming request came in =>
SIP Incoming Request: [[BYE 
sip:2406506175@20.20.20.20:5060<http://sip:2406506175@20.20.20.20:5060>;alias=10.10.10.5~55157~2
 SIP/2.0

I called handle_ruri_alias(), and expected the destination and port to be set 
to 10.10.10.5:55157<http://10.10.10.5:55157>, but it was not.
It was left 20.20.20.20:5060<http://20.20.20.20:5060>.

Is my thinking correct or is there another way to set the destination and port 
to the alias?

Thanks All!,
-Steve
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[SR-Users] handle_ruri_alias() question or issue?

[Wilkins, Steve]

Good Morning All,

The following Incoming request came in =>
SIP Incoming Request: [[BYE 
sip:2406506175@20.20.20.20:5060;alias=10.10.10.5~55157~2 SIP/2.0

I called handle_ruri_alias(), and expected the destination and port to be set 
to 10.10.10.5:55157, but it was not.
It was left 20.20.20.20:5060.

Is my thinking correct or is there another way to set the destination and port 
to the alias?

Thanks All!,
-Steve
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[SR-Users] Request Line alases

[Wilkins, Steve]

Hello All,


If the below Requests from 40.40.40.40(Asterisk) is received by 
10.10.10.10(Kamailio)
Request Line: INFO sip:3145553313@10.10.10.10:5060;alias=30.30.30.30~43508~2 
SIP/2.0

Shouldn't Kamailio forward this request to 30.30.30.30? so that the '200 OK' 
can sent back to Asterisk for the response.
(In this scenario) Would it be ok to just force a '200 OK' back to Asterisk 
from Kamailio?

Thank you,
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[SR-Users] Interesting issue with ACK back from 200 OK INVITE

[Wilkins, Steve]

Hello All,

I have had an issue for quite some time where I needed to open two UDP ports 
(via LISTEN) for some softphone UAC's to stay connected. Funny enough there is 
one Provider that works with the Single UDP.  I think I finally see why I was 
needing two listening UDP ports.  When the ACK is returning from Asterisk to 
Kamailio, the ACK is coming back as protocol TCP, not SIP, causing the call to 
drop.  I finally realized this but I am not sure what would cause this and why 
one provider phone works but not the others.  Once this is resolved, I may 
actually say...It works!

Thank you,
-Steve
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Re: [SR-Users] regular expression in kamailio

[Wilkins, Steve]

Okay, I will work with it some more.  been working since 6AM, maybe I just need 
to pick it up tomorrow.

Thanks again for your patience and help!
-Steve

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Saturday, August 25, 2018 7:34 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] regular expression in kamailio

On Sat, Aug 25, 2018 at 11:32:21PM +, Wilkins, Steve wrote:

> Sorry, bad habit of adding // to everything after :, and it was just example 
> of wanting to get the 10d number out of a string.  I will be more concise.
>   
> But here is what I am talking about.  This is  from a tcpdump trace.
> 
> From: "Test" 
> ;tag=1c993qq2-j4d1-411f-a7c9-11065fce48a2

$fU should work for this...

-- 
Alex Balashov | Principal | Evariste Systems LLC

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

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Re: [SR-Users] regular expression in kamailio

[Wilkins, Steve]

Sorry, bad habit of adding // to everything after :, and it was just example of 
wanting to get the 10d number out of a string.  I will be more concise.
  
But here is what I am talking about.  This is  from a tcpdump trace.

From: "Test" 
;tag=1c993qq2-j4d1-411f-a7c9-11065fce48a2

Thanks Alex,
-Steve

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Saturday, August 25, 2018 7:22 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] regular expression in kamailio

On Sat, Aug 25, 2018 at 11:16:29PM +0000, Wilkins, Steve wrote:

> Oh, but I want the 10 digit number. If from is like 
> sip://1@222333, I want 222333, not 1

The (abridged) grammar of a SIP URI is:

  sip:user@domain[:port]

The calling party number would be in the user part, ordinarily.

If you really want the domain, you're going to get a host/IP address of the 
calling UAC most likely. That can be accessed via @fd.

Perhaps you can provide an example of the exact From header you see and the 
portion of it you would like to extract? Because your example makes no sense. 
:-)

-- Alex

--
Alex Balashov | Principal | Evariste Systems LLC

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

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Re: [SR-Users] regular expression in kamailio

[Wilkins, Steve]

Oh, but I want the 10 digit number. If from is like
sip://1@222333, I want 222333, not 1

-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Saturday, August 25, 2018 7:11 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] regular expression in kamailio

On Sat, Aug 25, 2018 at 11:07:26PM +, Wilkins, Steve wrote:

> I was thinking that would also give me the IP or FQDN.  I should not 
> have assumed that.  I will go try it now.

Ah, no, that's $fu. :-)

Another option: $(fu{nameaddr.uri}{uri.user})

Now, of course, if you are not entirely sure what the format of $fU will be in 
any given scenario, there are a variety of things you can do to normalise it...

-- Alex

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

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Re: [SR-Users] regular expression in kamailio

[Wilkins, Steve]

Hi Alex,

I was thinking that would also give me the IP or FQDN.  I should not have 
assumed that.  I will go try it now.

Thank you!
-Steve
-Original Message-
From: sr-users  On Behalf Of Alex Balashov
Sent: Saturday, August 25, 2018 6:51 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] regular expression in kamailio

Why not just use $fU?

On Sat, Aug 25, 2018 at 10:46:36PM +, Wilkins, Steve wrote:

> Hi All,

> 
> I am trying get the 10digit number called in on using the following
> $var(caller) = 
> $(fu{re.subst,[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]});
> 
> I thought I could use a regular expression for the expression in 
> re.subst,expression
> 
> Thank you,
> -Steve

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

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[SR-Users] regular expression in kamailio

[Wilkins, Steve]

Hi All,

I am trying get the 10digit number called in on using the following
$var(caller) = 
$(fu{re.subst,[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]});

I thought I could use a regular expression for the expression in 
re.subst,expression

Thank you,
-Steve
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Re: [SR-Users] Listner problems

[Wilkins, Steve]

Hi Joel, and thank you for your response.

Actually, I use one IP but two different ports.  I have not yet figured out why 
some Providers need two listening ports but other need one.  As I mentioned, 
the ones that need one port will only have one way Media if two ports are used. 
 This is extremely Strang!

Yes, I am trying to use RTPEngine but have not been successful.  I have earlier 
posts (this week) which explains the problems I am having with RTPEngine.

Thanks again!,
-Steve

-Original Message-
From: sr-users  On Behalf Of Joel Serrano
Sent: Friday, August 24, 2018 7:04 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Listner problems

Hi Steve,

> Actually, I have a strange issue. If I have a single UDP IP:Port listening, 
> most Providers/Phones have two-way Audio/Video, however, with some 
> Providers/Phones, I need to have two UDP IP:Port listners in order to get 
> two-way Audio/Video.  Here is the strange thing.  If I enable two, the 
> Providers/Phones that worked when I had a single listener only have one-way 
> Audi/Video.
>

Can you give a little more details on why you need 2 IPs for some 
providers/phones?


> I use Kamailio 5.1 and Asterisk 15.3 (pjsip).  This behavior is so 
> strange,
>

Are you proxying RTP through Kamailio (rtpengine/rtpproxy) or is it going 
directly from the carrier<->asterisk?

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[SR-Users] Listner problems

[Wilkins, Steve]

Hi Joel,

Actually, I have a strange issue. If I have a single UDP IP:Port listening, 
most Providers/Phones have two-way Audio/Video, however, with some 
Providers/Phones, I need to have two UDP IP:Port listners in order to get 
two-way Audio/Video.  Here is the strange thing.  If I enable two, the 
Providers/Phones that worked when I had a single listener only have one-way 
Audi/Video.

I use Kamailio 5.1 and Asterisk 15.3 (pjsip).  This behavior is so strange,

Thank you,
-Steve

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Re: [SR-Users] add listen dynamically

[Wilkins, Steve]

Hi Joel,

Actually, I have a strange issue. If I have a single UDP IP:Port listening, 
most Providers/Phones have two-way Audio/Video, however, with some 
Providers/Phones, I need to have two UDP IP:Port listners in order to get 
two-way Audio/Video.  Here is the strange thing.  If I enable two, the 
Providers/Phones that worked when I had a single listener only have one-way 
Audi/Video.

I use Kamailio 5.1 and Asterisk 15.3 (pjsip).  This behavior is so strange,

Thank you,
-Steve

From: sr-users  On Behalf Of Joel Serrano
Sent: Friday, August 24, 2018 2:17 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] add listen dynamically

If you know the ips in advance you can specify all listen= params and enable 
the kernel ipv4_nonlocal_bind

Another option would be to add listen=* and have kamailio listen on anything 
the OS has enabled..?

Have you tried these couple options? Not sure if they would work without more 
details of what you are trying to achieve.



On Fri, Aug 24, 2018 at 10:35 Wilkins, Steve 
mailto:swwilk...@mitre.org>> wrote:
Hi All,

Is it possible to add or remove “listen” dynamically?

Thank you,
-Steve
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[SR-Users] add listen dynamically

[Wilkins, Steve]

Hi All,

Is it possible to add or remove "listen" dynamically?

Thank you,
-Steve
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[SR-Users] Struggling with RTPEngine

[Wilkins, Steve]

Hi Florian,

Thank you for your response.  I checked and direct-media is off.  Just as a 
recap, here is where I am

I just can't get rtpengine to work.  I have tried multiple configurations, but 
to no avail. Note that calls work good if rtpengine is disabled.

Here is my setup =>

Public IP: 20.20.20.20
Private IP 10.10.10.10

flow =>
webrtc client <-> kamailio+rtpengine <-> asterisk <-> kamailio <-> legacy sip 
phone

rtpenngine startup (I have tried a few different startups) =>
rtpengine --interface=int/10.10.10.10 --interface=ext/10.10.10.10\!20.20.20.20 
--listen-ng=127.0.0.1:12221 --pidfile=/var/run/rtpengine --dtls-passive -f -m 
1 -M 2 -E

kamailio =>
Invites: rtpengine_manage("trust-address replace-origin 
replace-session-connection direction=ext direction=int ICE=remove RTP/AVP");
Reply's: rtpengine_manage("trust-address replace-origin 
replace-session-connection ICE=force RTP/SAVPF");

I have tried direction ext ext; and many other combinations, each producing its 
own incorrect behavior.

Thanks again,
Steve

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Re: [SR-Users] Struggling with RTPProxy and RTPEngine

[Wilkins, Steve]

Good morning Daniel,

If RTPEngine is on the same server as Kamailio (Asterisk being on another 
server), and RTP traffic is sent to and from RTPEngine, then the provider only 
needs to whitelist one IP-Address.  I thought with RTPEngine that all RTP 
traffic would go through it and then it would pass it on to the correct 
destination.  Is this correct? 

Thank you

-Original Message-
From: sr-users  On Behalf Of Daniel Tryba
Sent: Thursday, August 23, 2018 4:36 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine

MITRE WARNING: Do not open unexpected password-protected attachments.

>>>Email originates from a non-MITRE system. Use caution.<<<

On Wed, Aug 22, 2018 at 05:05:02PM +0000, Wilkins, Steve wrote:
> The SIP traffic is working this way for me but I still see RTP traffic going 
> directly from Asterisk to the UAC, which means they need to whitelist 
> asterisk IP.  Am I missing something?

In what sense do they need whitelisting? In a common NATed solution where is no 
white/blacklist needed. UA gets RTP endpoints from SDP, starts sending packets 
to ip/port and the destination will send back packets to the source ip/port, 
the router/firewall will just send this to the actual UA. I have yet to find an 
UA that cares about where the RTP stream is coming from with regards to the SIP 
traffic.


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[SR-Users] RTPEngine wow's

[Wilkins, Steve]

Hello all,

I am still trying to get RTPEngine to work.  At this point, when I make a call, 
I do see the offer in the rtpengine log, I also see the update of the SDP (c) 
changing it to 20.20.20.20. However, after that, there are no logs other than a 
bunch of "timer run time = 0.nnn sec", and the delete when I hang up the 
call.  There is no RTP traffic what so ever in the rtpengine log (however, the 
call is completely successful, other than the RTP traffic not going through 
RTPEngine).

Here is a breakdown of my configs  (I borrowed this format from Maxim Fedorov 
because I thought it was very concise)

Starting rtpengine =>
rtpengine 
--interface=internal/21.21.21.21--interface=external/21.21.21.21\!20.20.20.20 
--listen-ng=127.0.0.1:12221 --dtls-passive -f -m 1 -M 2  -E -L 7 
--log-facility=local1

RTPEngine Public IP: 20.20.20.20
RTPEngine Private IP: 21.21.21.21

kamailio.cfg
modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:12221")
listen=tcp: 21.21.21.21:5060 advertise 20.20.20.20:5060
listen=udp: 21.21.21.21:5060 advertise 20.20.20.20:5060

rtpengine_manage call
rtpengine_manage("trust-address replace-origin replace-session-connection 
external external media-address=20.20.20.20"); (I have placed this 
rtpengine_manage call at the beginning of all routes).

I have tried many variations of rtpengine_manage parameters, and placement of 
the call itself, but I never see RTP traffic in the rtpengine log.

Thank you to all who have looked at this,
-Steve


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Re: [SR-Users] Struggling with RTPProxy and RTPEngine

[Wilkins, Steve]

Hello,

Yes, I don’t the UAC to have to whitelist the Asterisk Server.   Everything 
works great for SIP traffic and I want the same thing for RTP traffic, but I 
just can’t get rtpengine to work (and I have tried many configurations).
From: sr-users  On Behalf Of Sergiu Pojoga
Sent: Wednesday, August 22, 2018 11:23 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine

Is there a particular reason why you'd want to involve an rtp proxy in your 
setup?
Considering Asterisk is facing a public interface, media can be set up to flow 
direct between UACs and Asterisk.


On Wed, Aug 22, 2018 at 9:57 AM Daniel Tryba 
mailto:d.tr...@pocos.nl>> wrote:
On Wed, Aug 22, 2018 at 12:49:54PM +, Wilkins, Steve wrote:
> My offer and answer =>
> rtpengine_offer("trust-address replace-session-connection replace-origin");

If this is really your config you should change the offer to answer for
the answer part of the config. Either use rtpengine_offer and
rtengine_answer in the correct places of simply use rptengine_manage and
let the module figure out the right command.

I.O.W. you aren't giving any indepth info on your config.
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Re: [SR-Users] Struggling with RTPProxy and RTPEngine

[Wilkins, Steve]

I was attempting to log to local1 => rtpengine --interface 
172.21.1.108\!34.226.187.61 --listen-ng 127.0.0.1:12221 --dtls-passive -f -m 
1 -M 2  -E -L 7 --log-facility=local1.  However, even after adding 
"local1.*/var/log/rtpengine.log" to /etc/rsyslog.conf and 
restarting rsyslog, I get no logs.  I am on Cento7.  I do this for Kamailio and 
logging works.

Thank you for your response,
-Steve

From: sr-users  On Behalf Of Richard Fuchs
Sent: Wednesday, August 22, 2018 8:57 AM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine

It may be more helpful to post some logs from rtpengine. You should never see 
"Call-ID not found" from an offer.

Cheers

On 2018-08-22 08:49, Wilkins, Steve wrote:
Here is my start up =>
rtpengine --interface 111.121.22.11\!27.22.132.10 --listen-ng 127.0.0.1:12221 
--dtls-passive -f -m 1 -M 2  -E -L 7 --log-facility=local1

My offer and answer =>
rtpengine_offer("trust-address replace-session-connection replace-origin");

From: sr-users 
<mailto:sr-users-boun...@lists.kamailio.org>
 On Behalf Of Wilkins, Steve
Sent: Wednesday, August 22, 2018 8:43 AM
To: Kamailio (SER) - Users Mailing List 
<mailto:sr-users@lists.kamailio.org>
Subject: [SR-Users] Struggling with RTPProxy and RTPEngine

Hello all,

I can't seem to get either RTPProxy or RTPEngine to work correctly.  I have 
decided to concentrate on RTPEngine because I have read that it works the best. 
I am using Asterisk with Kamailio in front.  When I make calls, I see RTPEngine 
being hit but I continually get the error "Call-ID not found", calls work but 
media traffic doesn't go through RTPEngine.  I have tried many combination of 
the flags but I seem to always get the same error.When I look at the 
tcpdump log, I see that the media offer is not on the RTPEngine port. Is there 
a common mis-configuration error that can cause this?

I know I can send all my logs and configurations but I really want to try and 
resolve this as a learning experience.

Thanks all,
-Steve




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Re: [SR-Users] Struggling with RTPProxy and RTPEngine

[Wilkins, Steve]

Yes I do.

From: sr-users  On Behalf Of Nicolas Breuer
Sent: Wednesday, August 22, 2018 8:55 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine

Do you load the module ?


De : sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 De la part de Wilkins, Steve
Envoyé : mercredi 22 août 2018 14:50
À : Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Objet : Re: [SR-Users] Struggling with RTPProxy and RTPEngine

Here is my start up =>
rtpengine --interface 111.121.22.11\!27.22.132.10 --listen-ng 127.0.0.1:12221 
--dtls-passive -f -m 1 -M 2  -E -L 7 --log-facility=local1

My offer and answer =>
rtpengine_offer("trust-address replace-session-connection replace-origin");

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: Wednesday, August 22, 2018 8:43 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Struggling with RTPProxy and RTPEngine

Hello all,

I can't seem to get either RTPProxy or RTPEngine to work correctly.  I have 
decided to concentrate on RTPEngine because I have read that it works the best. 
I am using Asterisk with Kamailio in front.  When I make calls, I see RTPEngine 
being hit but I continually get the error "Call-ID not found", calls work but 
media traffic doesn't go through RTPEngine.  I have tried many combination of 
the flags but I seem to always get the same error.When I look at the 
tcpdump log, I see that the media offer is not on the RTPEngine port. Is there 
a common mis-configuration error that can cause this?

I know I can send all my logs and configurations but I really want to try and 
resolve this as a learning experience.

Thanks all,
-Steve
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Re: [SR-Users] Struggling with RTPProxy and RTPEngine

[Wilkins, Steve]

Here is my start up =>
rtpengine --interface 111.121.22.11\!27.22.132.10 --listen-ng 127.0.0.1:12221 
--dtls-passive -f -m 1 -M 2  -E -L 7 --log-facility=local1

My offer and answer =>
rtpengine_offer("trust-address replace-session-connection replace-origin");

From: sr-users  On Behalf Of Wilkins, Steve
Sent: Wednesday, August 22, 2018 8:43 AM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Struggling with RTPProxy and RTPEngine

Hello all,

I can't seem to get either RTPProxy or RTPEngine to work correctly.  I have 
decided to concentrate on RTPEngine because I have read that it works the best. 
I am using Asterisk with Kamailio in front.  When I make calls, I see RTPEngine 
being hit but I continually get the error "Call-ID not found", calls work but 
media traffic doesn't go through RTPEngine.  I have tried many combination of 
the flags but I seem to always get the same error.When I look at the 
tcpdump log, I see that the media offer is not on the RTPEngine port. Is there 
a common mis-configuration error that can cause this?

I know I can send all my logs and configurations but I really want to try and 
resolve this as a learning experience.

Thanks all,
-Steve
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[SR-Users] Struggling with RTPProxy and RTPEngine

[Wilkins, Steve]

Hello all,

I can't seem to get either RTPProxy or RTPEngine to work correctly.  I have 
decided to concentrate on RTPEngine because I have read that it works the best. 
I am using Asterisk with Kamailio in front.  When I make calls, I see RTPEngine 
being hit but I continually get the error "Call-ID not found", calls work but 
media traffic doesn't go through RTPEngine.  I have tried many combination of 
the flags but I seem to always get the same error.When I look at the 
tcpdump log, I see that the media offer is not on the RTPEngine port. Is there 
a common mis-configuration error that can cause this?

I know I can send all my logs and configurations but I really want to try and 
resolve this as a learning experience.

Thanks all,
-Steve
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Re: [SR-Users] help

[Wilkins, Steve]

Hi Pravin,

I would start off by  doing an clean of your repositories, and then possibly 
re-installing MariaDB.  I also use MariaDB for my installation with Kamailio and
the only issue I remember having was some missing (.h) files, which I then had 
to go the develop libraries for MariaDB.

-Steve

From: sr-users  On Behalf Of Pravin .
Sent: Monday, August 20, 2018 9:52 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] help

Herewith attaching the screenshot for the error we are getting after issuing 
apt-get install mariadb-server command...

Pls have a look on attached file.

Regards,
Pravin

On Mon, Aug 20, 2018 at 6:58 PM, Pravin . 
mailto:pra...@bolindia.com>> wrote:
thanks for your support but the following command not worked properly..

1. apt-get install mariadb-server
after running this command it works but some depended packages not installed 
correctly..and then what will be next command as given below as it depends on 
earlier package installed ..

2. apt install kamailio kamailio-mysql-modules  (how it run as it depend on 
earlier package installed)

If you can provide us the latest /updated install guide for kamailio on debian, 
would be helpful.

Regards,
Pravin
bolindia Networks Pvt. Ltd.


On Mon, Aug 20, 2018 at 6:24 PM, Floimair Florian 
mailto:f.floim...@commend.com>> wrote:
MySQL is no longer part of Debian. Instead they now use the MySQL fork MariaDB.
So instead of
mysql-server use maria-db-server

all the rest of the steps are the same.



With best regards

Florian Floimair
Innovation - Software-Development

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

Security and Communication by Commend

FN 178618z | LG Salzburg

Von: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 im Auftrag von "Pravin ." mailto:pra...@bolindia.com>>
Antworten an: "Kamailio (SER) - Users Mailing List" 
mailto:sr-users@lists.kamailio.org>>
Datum: Montag, 20. August 2018 um 14:51
An: "Kamailio (SER) - Users Mailing List" 
mailto:sr-users@lists.kamailio.org>>
Betreff: Re: [SR-Users] help

Hello ,

Tried to install Kamailio as per link provided by you  
https://www.kamailio.org/wiki/install/stable/debian.

following commands are not working...

apt
install
mysql-server
apt install
kamailio kamailio-mysql-modules

Getting following error...

Package mysql-server is not available ,but is referred to by another package.
This may mean that package is missing.
E: Package "mysql-server" has no installation candidate

Pls guide us how to proceed...


Note: I have installed Debian 9 as OS in my system.

Regards,
Pravin
bolindia Networks Pvt Ltd.

On Sat, Aug 18, 2018 at 1:38 PM, Mojtaba 
mailto:mes...@gmail.com>> wrote:
Hi,
The installation of Kamailio is straight forward, Just follow the
following steps in this site:
https://www.kamailio.org/wiki/install/stable/debian
Also, you could download Kamailio form git source.
With Regards.Mojtaba

On Sat, Aug 18, 2018 at 11:52 AM Pravin . 
mailto:pra...@bolindia.com>> wrote:
>
> Hello Team,
>
> I want to install Kamailio SIP server on centOS server...do we need to 
> download kamailio ISO file and install on server..pls provide the 
> installation procedure/guidelines.
>
>
> Regards
> Pravin
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--
--Mojtaba Esfandiari.S

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[SR-Users] Mysql 5.7 vs 8.0

[Wilkins, Steve]

Hi All,

I would like Kamailio to use MySQL 5.7, however when Kamailio installs and I 
say I want to use MySQL, it installs 8.0.  Can I someway direct Kamailio to 
install version 5.7?

Thank you,
-Steve
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Re: [SR-Users] kamailio rtp proxy set not working

[Wilkins, Steve]

Are your ports open?

-Original Message-
From: sr-users  On Behalf Of Henning 
Westerholt
Sent: Tuesday, August 14, 2018 1:08 PM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] kamailio rtp proxy set not working

Am Montag, 13. August 2018, 13:32:29 CEST schrieb ANOOP V M:
> I have installed kamailio 5.1.4 version in ubuntu server. I need to 
> route RTP packets through kamailio with SIP packets. I will explain 
> the way i have set up everything.
> [...]
> Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: ERR:create_twinlistener:
> can't bind to the IPv4 port 47926: Cannot assign requested address Aug 
> 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: ERR:handle_command: can't 
> create listener

Hello Anoop,

I am not an rtpproxy expert. But I would suggest that you look into the error 
quoted above. This means that rtproxy can't bind to the respective port and or 
address. There are several possible reasons for that:

- this IP is not configured (correctly) on the server
- the port is already allocated to another process
- there is some other error in network configuration

Best regards,

Henning

--
Henning Westerholt
https://skalatan.de/blog/

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[SR-Users] rtpproxy or rtpengine with Kamailio and Asterisk

[Wilkins, Steve]

Good Morning everyone,

Has anyone ever gotten rtpproxy/rtpengine to work with Kamailio and Asterisk?

I have tried both, and even though I see no errors with either, and when I make 
calls I see rtpproxy/rtpengine traffic, I am still seeing media traffic go 
directly to and from Asterisk.  I just can't get the media packets to not go 
directly from Asterisk to the Callee and visa-versa.

Any ideas on where I am messing up in my kamalio.cfg file?

Thank you!,
-Steve
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Re: [SR-Users] Rtpengine?

[Wilkins, Steve]

Hello,

I switched to rtpproxy;  rtpproxy runs with no errors, Kamailio runs with no 
errors, but Asterisk is still sending the media packets, I want the proxy to be 
sending the packets (this way, the providers have no knowledge of Asterisk and 
no need to whitelist every provider, just Kamailio).
Do I need to add rtpproxy_manage() at every SIP message?

Thank you,
-Steve

From: sr-users  On Behalf Of Yu Boot
Sent: Tuesday, August 14, 2018 2:12 AM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] Rtpengine?


Hello.

It's not enough to install rtpproxy/rtpengine and start it. You need to call 
rtpengine functions from kamailio.cfg for every SIP packet with SDP so media 
will be routed "thru Kamailio".

To make it easier, I recommend to begin with rtpproxy, not rtpengine. Default 
kamailio.cfg contains everything you need to proxy media flow by rtpproxy, just 
enable WITH_NAT parameter.

14.08.2018 1:51, Wilkins, Steve пишет:
HI All,

I am not sure if I understand it correctly but I thought that I could use 
rtpengine to redirect media packets.  My current SIP flow is =>
Softphone=>Kamailio=>Asterisk=>Kamailio=>Softphone, and Media flows from is 
Asterisk<=>softphone. But I don’t want Media to flow this way.
That is, I do not want Asterisk and the Softphone to be aware of each other. I 
would like the Media to go through Kamailio just like SIP packets do.
I installed rtpengine and it starts and runs with no errors.  I even see 
traffic with the call is made but media traffic still flows from 
softphone<=>asterisk.



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Re: [SR-Users] Rtpengine?

[Wilkins, Steve]

Hello and thank you for your response.  I was going on something I read that 
said =>

“Getting rtp_engine to work with the standard kamailio.cfg is to simply add
rtpengine_manage();
at the top of the NATMANAGE route so it will handle all rtp regardless of
source being natted (in your config in the NAT router.”

Obviously, this is not correct.

From: sr-users  On Behalf Of Yu Boot
Sent: Tuesday, August 14, 2018 2:12 AM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] Rtpengine?


Hello.

It's not enough to install rtpproxy/rtpengine and start it. You need to call 
rtpengine functions from kamailio.cfg for every SIP packet with SDP so media 
will be routed "thru Kamailio".

To make it easier, I recommend to begin with rtpproxy, not rtpengine. Default 
kamailio.cfg contains everything you need to proxy media flow by rtpproxy, just 
enable WITH_NAT parameter.

14.08.2018 1:51, Wilkins, Steve пишет:
HI All,

I am not sure if I understand it correctly but I thought that I could use 
rtpengine to redirect media packets.  My current SIP flow is =>
Softphone=>Kamailio=>Asterisk=>Kamailio=>Softphone, and Media flows from is 
Asterisk<=>softphone. But I don’t want Media to flow this way.
That is, I do not want Asterisk and the Softphone to be aware of each other. I 
would like the Media to go through Kamailio just like SIP packets do.
I installed rtpengine and it starts and runs with no errors.  I even see 
traffic with the call is made but media traffic still flows from 
softphone<=>asterisk.



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[SR-Users] Rtpengine?

[Wilkins, Steve]

HI All,

I am not sure if I understand it correctly but I thought that I could use 
rtpengine to redirect media packets.  My current SIP flow is =>
Softphone=>Kamailio=>Asterisk=>Kamailio=>Softphone, and Media flows from is 
Asterisk<=>softphone. But I don't want Media to flow this way.
That is, I do not want Asterisk and the Softphone to be aware of each other. I 
would like the Media to go through Kamailio just like SIP packets do.
I installed rtpengine and it starts and runs with no errors.  I even see 
traffic with the call is made but media traffic still flows from 
softphone<=>asterisk.

Any ideas?

Thank you,

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Re: [SR-Users] Migrate from RtpProxy to Rtpengine

[Wilkins, Steve]

Are you going through a PBX like Asterisk?  I am using rtpengine but I cannot 
get media packets to go from Asterisk->Kamailio(rtpengine)->softphone.  I get 
no errors and I see rtpengine traffic, but the calls go Asterisk->softphone.

Thanks ALL

From: sr-users  On Behalf Of Nicolas Breuer
Sent: Monday, August 13, 2018 11:32 AM
To: 'Kamailio (SER) - Users Mailing List' 
Subject: [SR-Users] Migrate from RtpProxy to Rtpengine

Hello,

I migrated from Rtp_proxy module to Rtp_Engine.
Everythings works fine apart the rewriting of the SDP description in Kamailio

subst_body('/(^s=.*)/s=Test of a Sip call.\r/')

Any ideas if this is supposed to be supported?


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[SR-Users] Can Kamailio interface with an asterisk server based on the fqdn of a call?

[Wilkins, Steve]

Hi All,

Is it possible for Kamailio to interface with a particular Asterisk Server 
based on the FQDN of a caller?

I would like to pass calls received by Kamailio through to different Asterisk 
Servers based on the FQDN of the caller.  I have used Load Balancing before, 
but I want to select which Asterisk Server the call is directed to.

Thank you,

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[SR-Users] "kamdbctl create" failing with MySQL 8

[Wilkins, Steve]

Hello All,

There appears to be issues with running "kamdbctl create" when using MySQL 8.  
When this is ran there are syntax SQL errors.  One such error is when doing 
grants using  "IDENTIFIED BY 'password'";  This throws a sequel error for 
version 8 of MySQL.

Thank you,
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Re: [SR-Users] Other issues encountered with MySQL 8 and Kamailio

[Wilkins, Steve]

This does not happen if I am using MariaDB.  It appears to be either a MySQL 8 
issue or MySQL issue in general.

From: sr-users  On Behalf Of Wilkins, Steve
Sent: Friday, August 3, 2018 6:53 AM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Other issues encountered with MySQL 8 and Kamailio

Good Morning all,

When I do get Kamailio to compile with MySQL 8, I encounter the following issue 
when running "kamdbctl create"

ERROR 1449 (HY000) at line 1: The user specified as a definer 
('mysql.infoschema'@'localhost') does not exist
WARNING: Your current default mysql characters set cannot be used to create DB. 
Please choice another one from the following list:

Any ideas?

Thank you.
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[SR-Users] Other issues encountered with MySQL 8 and Kamailio

[Wilkins, Steve]

Good Morning all,

When I do get Kamailio to compile with MySQL 8, I encounter the following issue 
when running "kamdbctl create"

ERROR 1449 (HY000) at line 1: The user specified as a definer 
('mysql.infoschema'@'localhost') does not exist
WARNING: Your current default mysql characters set cannot be used to create DB. 
Please choice another one from the following list:

Any ideas?

Thank you.
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[SR-Users] 5.1 compilation error on Centos 7

[Wilkins, Steve]

Hello all,

I started getting the following error while trying to compile Kamailio 5.1+ on 
Centos 7

CC (gcc) [M db_mysql.so]my_fld.o
In file included from my_fld.c:22:0:
my_fld.h:37:2: error: unknown type name 'my_bool'
  my_bool is_null;

Has anyone seen this before.

Thank you
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[SR-Users] Difference between allow trusted and allow address

[Wilkins, Steve]

Hi All,

Can someone explain the difference between allow trusted and allow address.

Thank you!
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Re: [SR-Users] Kamailio as outbound proxy for PBX

[Wilkins, Steve]

Current Outbound Call:
WebRTC Client => Asterisk =>SIPDevice, here the SIPDevice communicates back 
through Asterisk.

Desired Outbound Call:
WebRTC Client => Asterisk (which has an outbound_proxy set in pjsip) 
=>Kamailio=>SIPDevice and back the same way?  
The end goal being that the SIPDevice never sees Asterisk.

I have attempted doing the Desired.  Asterisk is indeed sending the outbound 
calls to Kamailio, but Kamailio is not contacting the SIPDevice (number@fqdn).


Thank you,


-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Daniel 
Tryba
Sent: Thursday, June 14, 2018 11:40 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Kamailio as outbound proxy for PBX

On Thu, Jun 14, 2018 at 10:56:51AM +, Wilkins, Steve wrote:
> If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio 
> actually make the SIP call?

Ehhh, yes. Why wouldn't that be possible?
 
> At some point I would like outbound calls to be controlled by Kamailio 
> so that the outside endpoints never communicate with the PBX.
> Currently a call goes through Kamailio to Asterisk and then Asterisk 
> communicates with the endpoint, and this is night ideal

I don't see what kind of scenario you want to implement and what your current 
setup is.


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[SR-Users] Kamailio as outbound proxy for PBX

[Wilkins, Steve]

Good Morning All!

If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio 
actually make the SIP call?

At some point I would like outbound calls to be controlled by Kamailio so that 
the outside endpoints never communicate with the PBX.
Currently a call goes through Kamailio to Asterisk and then Asterisk 
communicates with the endpoint, and this is night ideal

Thanks ALL
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[SR-Users] Asterisk inviting Kamailio

[Wilkins, Steve]

Hello All,

I have noticed that sometimes when a call is made from one endpoint to another 
through Asterisk via Kamailio, Asterisk sends an INVITE to Kamailio even after 
the call has been established.  Sometimes this does not happen.   When it does 
happen, calls drop.
Why would an INVITE be sent back to Kamailio?

Also, Pan, I am working on the response you requested yesterday.

Thanks you All,
-Steve
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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio

[Wilkins, Steve]

Hi Pan,

I will type something up.  It may take me a bit to explain it correctly, but I 
will definitely send you something.  Although I now have a problem with 
outbound calls hanging up, I at least get two-way Audio and Video.  I think the 
problem is just moving around.

Thank you,
-Steve

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Tuesday, June 12, 2018 8:18 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using 
Kamailio...works without Kamailio

Dear Steve,

Would you mind sharing your findings and solution with the list?

With kind regards
Pan B. Christensen
Developer
Phonect AS 

> -Original Message-
> From: sr-users  On Behalf Of 
> Wilkins, Steve
> Sent: mandag 11. juni 2018 12:30
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone 
> when using Kamailio...works without Kamailio
> 
> Got it working.  Thank you everyone.
> 
> -Original Message-
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf 
> Of Wilkins, Steve
> Sent: Sunday, June 10, 2018 3:06 PM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone 
> when using Kamailio...works without Kamailio
> 
> >>>Email originates from a non-MITRE system. Use caution.<<<
> 
> Alex, Pan, Daniel,...
> Could this group => group:BUNDLE audio video in Message Body have 
> anything to do with my Kamailio Video issue.
> 
> Thank you!!
> -Steve
> 
> -Original Message-
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf 
> Of Wilkins, Steve
> Sent: Sunday, June 10, 2018 12:14 PM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone 
> when using Kamailio...works without Kamailio
> 
> Hi Alex,
> No, I'm not using rtpengine.  It must definitely be some sort of Codec 
> issue though, since I can seem to move the problem around and actually 
> finally get video on the  softphone.  It was a little strange that I 
> lost Audio on the WebRTC client though, since I only disabled VP8.
> 
> Since Kamailio is only relaying I am so confused why introducing 
> Kamailio is messing up Video.
> 
> Thank you,
> -Steve
> 
> -Original Message-
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf 
> Of Alex Balashov
> Sent: Sunday, June 10, 2018 11:46 AM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone 
> when using Kamailio...works without Kamailio
> 
> Unless you're using rtpengine, you're just moving the problem around.
> Kamailio does nothing with SDP unless by way of rtpengine or told in 
> some other way, such as sdpops.
> 
> On June 10, 2018 11:44:47 AM EDT, "Wilkins, Steve" 
> 
> wrote:
> >I just did a test where I disabled VP8 in Kamailio using SDPOPS and I 
> >now get Video on the softphone however, I lost two-way Audio.  
> >Kamailio seems to be  doing something with the codecs but still can't 
> >put my finger on it.  The WebRTC Client, who is the caller, needs VP8 
> >for Video, and apparently Audio.
> >
> >I'm not sure if I getting closer or just moving the problem around.
> >
> >Thanks All!
> >
> >From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf 
> >Of Wilkins, Steve
> >Sent: Saturday, June 9, 2018 12:47 PM
> >To: Kamailio (SER) - Users Mailing List 
> >Subject: [SR-Users] No Video between WebRTC Client and Softphone when 
> >using Kamailio...works without Kamailio
> >
> >>>>Email originates from a non-MITRE system. Use caution.<<<
> >I am desperately trying to resolve a big issue I have when using 
> >Kamailio 5.2.0 and Asterisk 15.3.
> >
> >As soon as I go through Kamailio, I get no Video on either side of 
> >the call; I do get two-way audio and the call stays connected.  If I 
> >simply go right through Asterisk, I also get two-way Video.  I am 
> >having a tough time determining why Kamailio is messing with the 
> >Video portion of the call.
> >I have included a full pcap file.  As a side note when comparing pcap 
> >traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from 
> >Asterisk and Softphone are exactly the same.
> >
> >Thank you,
> >-Steve
> 
> 
> -- Alex
> 
> --
> Sent via mobile, please forgive typos and brevity.
> 
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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio

[Wilkins, Steve]

Got it working.  Thank you everyone.

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Wilkins, Steve
Sent: Sunday, June 10, 2018 3:06 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using 
Kamailio...works without Kamailio

>>>Email originates from a non-MITRE system. Use caution.<<<

Alex, Pan, Daniel,...
Could this group => group:BUNDLE audio video in Message Body have anything to 
do with my Kamailio Video issue.

Thank you!!
-Steve

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Wilkins, Steve
Sent: Sunday, June 10, 2018 12:14 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using 
Kamailio...works without Kamailio

Hi Alex,
No, I'm not using rtpengine.  It must definitely be some sort of Codec issue 
though, since I can seem to move the problem around and actually finally get 
video on the  softphone.  It was a little strange that I lost Audio on the 
WebRTC client though, since I only disabled VP8.

Since Kamailio is only relaying I am so confused why introducing Kamailio is 
messing up Video.

Thank you,
-Steve

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Alex 
Balashov
Sent: Sunday, June 10, 2018 11:46 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using 
Kamailio...works without Kamailio

Unless you're using rtpengine, you're just moving the problem around. Kamailio 
does nothing with SDP unless by way of rtpengine or told in some other way, 
such as sdpops. 

On June 10, 2018 11:44:47 AM EDT, "Wilkins, Steve"  wrote:
>I just did a test where I disabled VP8 in Kamailio using SDPOPS and I 
>now get Video on the softphone however, I lost two-way Audio.  Kamailio 
>seems to be  doing something with the codecs but still can't put my 
>finger on it.  The WebRTC Client, who is the caller, needs VP8 for 
>Video, and apparently Audio.
>
>I'm not sure if I getting closer or just moving the problem around.
>
>Thanks All!
>
>From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf 
>Of Wilkins, Steve
>Sent: Saturday, June 9, 2018 12:47 PM
>To: Kamailio (SER) - Users Mailing List 
>Subject: [SR-Users] No Video between WebRTC Client and Softphone when 
>using Kamailio...works without Kamailio
>
>>>>Email originates from a non-MITRE system. Use caution.<<<
>I am desperately trying to resolve a big issue I have when using 
>Kamailio 5.2.0 and Asterisk 15.3.
>
>As soon as I go through Kamailio, I get no Video on either side of the 
>call; I do get two-way audio and the call stays connected.  If I simply 
>go right through Asterisk, I also get two-way Video.  I am having a 
>tough time determining why Kamailio is messing with the Video portion 
>of the call.
>I have included a full pcap file.  As a side note when comparing pcap 
>traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from 
>Asterisk and Softphone are exactly the same.
>
>Thank you,
>-Steve


-- Alex

--
Sent via mobile, please forgive typos and brevity. 

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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio

[Wilkins, Steve]

Alex, Pan, Daniel,...
Could this group => group:BUNDLE audio video in Message Body have anything to 
do with my Kamailio Video issue.

Thank you!!
-Steve

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Wilkins, Steve
Sent: Sunday, June 10, 2018 12:14 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using 
Kamailio...works without Kamailio

Hi Alex,
No, I'm not using rtpengine.  It must definitely be some sort of Codec issue 
though, since I can seem to move the problem around and actually finally get 
video on the  softphone.  It was a little strange that I lost Audio on the 
WebRTC client though, since I only disabled VP8.

Since Kamailio is only relaying I am so confused why introducing Kamailio is 
messing up Video.

Thank you,
-Steve

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Alex 
Balashov
Sent: Sunday, June 10, 2018 11:46 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using 
Kamailio...works without Kamailio

Unless you're using rtpengine, you're just moving the problem around. Kamailio 
does nothing with SDP unless by way of rtpengine or told in some other way, 
such as sdpops. 

On June 10, 2018 11:44:47 AM EDT, "Wilkins, Steve"  wrote:
>I just did a test where I disabled VP8 in Kamailio using SDPOPS and I 
>now get Video on the softphone however, I lost two-way Audio.  Kamailio 
>seems to be  doing something with the codecs but still can't put my 
>finger on it.  The WebRTC Client, who is the caller, needs VP8 for 
>Video, and apparently Audio.
>
>I'm not sure if I getting closer or just moving the problem around.
>
>Thanks All!
>
>From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf 
>Of Wilkins, Steve
>Sent: Saturday, June 9, 2018 12:47 PM
>To: Kamailio (SER) - Users Mailing List 
>Subject: [SR-Users] No Video between WebRTC Client and Softphone when 
>using Kamailio...works without Kamailio
>
>>>>Email originates from a non-MITRE system. Use caution.<<<
>I am desperately trying to resolve a big issue I have when using 
>Kamailio 5.2.0 and Asterisk 15.3.
>
>As soon as I go through Kamailio, I get no Video on either side of the 
>call; I do get two-way audio and the call stays connected.  If I simply 
>go right through Asterisk, I also get two-way Video.  I am having a 
>tough time determining why Kamailio is messing with the Video portion 
>of the call.
>I have included a full pcap file.  As a side note when comparing pcap 
>traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from 
>Asterisk and Softphone are exactly the same.
>
>Thank you,
>-Steve


-- Alex

--
Sent via mobile, please forgive typos and brevity. 

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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio

[Wilkins, Steve]

Hi Alex,
No, I'm not using rtpengine.  It must definitely be some sort of Codec issue 
though, since I can seem to move the problem around and actually finally get 
video on the  softphone.  It was a little strange that I lost Audio on the 
WebRTC client though, since I only disabled VP8.

Since Kamailio is only relaying I am so confused why introducing Kamailio is 
messing up Video.

Thank you,
-Steve

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Alex 
Balashov
Sent: Sunday, June 10, 2018 11:46 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using 
Kamailio...works without Kamailio

Unless you're using rtpengine, you're just moving the problem around. Kamailio 
does nothing with SDP unless by way of rtpengine or told in some other way, 
such as sdpops. 

On June 10, 2018 11:44:47 AM EDT, "Wilkins, Steve"  wrote:
>I just did a test where I disabled VP8 in Kamailio using SDPOPS and I 
>now get Video on the softphone however, I lost two-way Audio.  Kamailio 
>seems to be  doing something with the codecs but still can't put my 
>finger on it.  The WebRTC Client, who is the caller, needs VP8 for 
>Video, and apparently Audio.
>
>I'm not sure if I getting closer or just moving the problem around.
>
>Thanks All!
>
>From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf 
>Of Wilkins, Steve
>Sent: Saturday, June 9, 2018 12:47 PM
>To: Kamailio (SER) - Users Mailing List 
>Subject: [SR-Users] No Video between WebRTC Client and Softphone when 
>using Kamailio...works without Kamailio
>
>>>>Email originates from a non-MITRE system. Use caution.<<<
>I am desperately trying to resolve a big issue I have when using 
>Kamailio 5.2.0 and Asterisk 15.3.
>
>As soon as I go through Kamailio, I get no Video on either side of the 
>call; I do get two-way audio and the call stays connected.  If I simply 
>go right through Asterisk, I also get two-way Video.  I am having a 
>tough time determining why Kamailio is messing with the Video portion 
>of the call.
>I have included a full pcap file.  As a side note when comparing pcap 
>traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from 
>Asterisk and Softphone are exactly the same.
>
>Thank you,
>-Steve


-- Alex

--
Sent via mobile, please forgive typos and brevity. 

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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio

[Wilkins, Steve]

I just did a test where I disabled VP8 in Kamailio using SDPOPS and I now get 
Video on the softphone however, I lost two-way Audio.  Kamailio seems to be  
doing something with the codecs but still can't put my finger on it.  The 
WebRTC Client, who is the caller, needs VP8 for Video, and apparently Audio.

I'm not sure if I getting closer or just moving the problem around.

Thanks All!

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Wilkins, Steve
Sent: Saturday, June 9, 2018 12:47 PM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] No Video between WebRTC Client and Softphone when using 
Kamailio...works without Kamailio

>>>Email originates from a non-MITRE system. Use caution.<<<
I am desperately trying to resolve a big issue I have when using Kamailio 5.2.0 
and Asterisk 15.3.

As soon as I go through Kamailio, I get no Video on either side of the call; I 
do get two-way audio and the call stays connected.  If I simply go right 
through Asterisk, I also get two-way Video.  I am having a tough time 
determining why Kamailio is messing with the Video portion of the call.
I have included a full pcap file.  As a side note when comparing pcap traces of 
the Non Proxied vs Proxied call, the INVITE and 200 OK from Asterisk and 
Softphone are exactly the same.

Thank you,
-Steve
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Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Wilkins, Steve]

Pan,
I did verify the '200 OK' is making it to the softphone because I see the 'ACK' 
to it.  So that isn't the issue.

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Saturday, June 9, 2018 6:20 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

I have to admit that I don't have much experience with video calls, but as far 
as I can tell VP8 is negotiated with WebRTC client. Nothing is negotiated with 
softphone.

WebRTC client offers:
Media Description, name and address (m): video 55185 UDP/TLS/RTP/SAVPF 96 97 98 
99 100 101 102 123 127 122 125 107 108 109 124

Asterisk offers to softphone:
Media Description, name and address (m): video 15112 RTP/AVP 99
Why is it advertising only 99?

Softphone answers with:
Media Description, name and address (m): video 36140 RTP/AVP 115 113 34 31
No match with offer, video is not negotiated for this call leg.

Asterisk answers WebRTC with:
Media Description, name and address (m): video 10328 UDP/TLS/RTP/SAVPF 96 100
96 is chosen for this call leg.


With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: lørdag 9. juni 2018 03:59
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello again Pan and Thank You for looking at this!!  Here is the pcap for the 
call that is not showing video on either side of the call.  As I mentioned I do 
get two way Audio, the call stays connected, but just no Video.  The Video 
ports look to be correct.  Video packets appear to go out (at least packets are 
going out over the Video ports). The screen on the Caller (A WebRTC Client) and 
the Called (A Provider Phone) are both blank.  The WebRTC client is Registered 
in Kamailio

Note: when I sanitized the file using tcprewrite an extra RTP packet appeared 
(Unknown RTP packet version 0), this was not here in the original.

I use Kamailio 5.2.0 and Asterisk 15.3

Thank you again!,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 5:12 PM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Dear Steve.

Asterisk is a B2BUA. It doesn't just route SIP messages like Kamailio does. 
Most of the time, it answers the call, sets up a new call to the other party 
and then bridges the two calls, making two separate call IDs (depending on your 
configuration).

Additional comments in blue inline below:


  *   INVITE from Caller to Kamailio (this is where the Caller is Registered) 
offering VP8
  *   Kamailio will relay the INVITE to Asterisk
  * Asterisk probably sends 183/200 back to callee, finalizing this codec 
negotiation. VP8 is chosen for this call leg.
  *   Asterisk will INVITE Called
  *   Called sends '200 OK' to Asterisk requesting H264
  * yes
  *   Asterisk Sends '200 OK' back to Kamailio to be relayed to Caller 
accepting VP8
  * No. I assume that the callee didn't accept VP8 because it doesn't 
support it. It instead chose the highest priority codec it supported among the 
codecs that Asterisk advertised.
  * Asterisk accepts this codec because it supports it. There is no test 
checking which codec was chosen on the caller side and there is no 
renegotiation. Hence, video is broken forever.

Again, if you could show us the details (like attaching the SDPs that I asked 
for previously), then we wouldn't have to guess what's happening. We could tell 
you for sure.


Med vennlig hilsen
Pan B. Christensen

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 22:27
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi Pan,

If the Registered caller is a WebRTC client (Caller) whose preferred coded is 
VP8, calls a Phone (Called) whose preferred Video is H264, I should see


  *   INVITE from Caller to Kamailio (this is where the Caller is Registered) 
offering VP8
  *   Kamailio will relay the INVITE to Asterisk
  *   Asterisk will INVITE Called
  *   Called sends '200 OK' to Asterisk requesting H264
  *   Asterisk Sends '200 OK' back to Kamailio to be relayed to Caller 
accepting VP8

Is this correct?  Then I think one of the issues is that there is no fmtp line 
in the VP8.  The only codecs that have fmtp lines is for the H264 codecs.

Thank you,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 11:06 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [

Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Wilkins, Steve]

Hi Pan,

I once again compared the call through the Proxy and without the Proxy; the 
Offers appear to be exactly the same, so I think the negotiation must be 
working?, even though one is 99 and one is 115 does not appear to matter as the 
call just through Asterisk works.  I just can't figure out why the Proxy has 
any effect.  Is there a chance that the '200 OK' is not being relayed to the 
softphone when going through the proxy?  I asked this because I see the '200 
OK' coming into Kamailio, but I'm not so sure it is getting relayed.  Is this 
correct thinking?

These are the same with or without Proxy =>

Asterisk (INVITE)
Media Description, name and address (m): video 15112 RTP/AVP 99
Media Attribute (a): rtpmap:99 H264/9
Media Attribute (a): fmtp:99 
level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f

Softphone (200 OK)
Media Description, name and address (m): video 36140 RTP/AVP 115 113 34 31
Media Attribute (a): rtpmap:115 H264/9
Media Attribute (a): fmtp:115 
profile-level-id=42000b;packetization-mode=1;max-mbps=23760

Thanks!!,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Saturday, June 9, 2018 6:20 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

I have to admit that I don't have much experience with video calls, but as far 
as I can tell VP8 is negotiated with WebRTC client. Nothing is negotiated with 
softphone.

WebRTC client offers:
Media Description, name and address (m): video 55185 UDP/TLS/RTP/SAVPF 96 97 98 
99 100 101 102 123 127 122 125 107 108 109 124

Asterisk offers to softphone:
Media Description, name and address (m): video 15112 RTP/AVP 99
Why is it advertising only 99?

Softphone answers with:
Media Description, name and address (m): video 36140 RTP/AVP 115 113 34 31
No match with offer, video is not negotiated for this call leg.

Asterisk answers WebRTC with:
Media Description, name and address (m): video 10328 UDP/TLS/RTP/SAVPF 96 100
96 is chosen for this call leg.


With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: lørdag 9. juni 2018 03:59
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello again Pan and Thank You for looking at this!!  Here is the pcap for the 
call that is not showing video on either side of the call.  As I mentioned I do 
get two way Audio, the call stays connected, but just no Video.  The Video 
ports look to be correct.  Video packets appear to go out (at least packets are 
going out over the Video ports). The screen on the Caller (A WebRTC Client) and 
the Called (A Provider Phone) are both blank.  The WebRTC client is Registered 
in Kamailio

Note: when I sanitized the file using tcprewrite an extra RTP packet appeared 
(Unknown RTP packet version 0), this was not here in the original.

I use Kamailio 5.2.0 and Asterisk 15.3

Thank you again!,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 5:12 PM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Dear Steve.

Asterisk is a B2BUA. It doesn't just route SIP messages like Kamailio does. 
Most of the time, it answers the call, sets up a new call to the other party 
and then bridges the two calls, making two separate call IDs (depending on your 
configuration).

Additional comments in blue inline below:


  *   INVITE from Caller to Kamailio (this is where the Caller is Registered) 
offering VP8
  *   Kamailio will relay the INVITE to Asterisk
  * Asterisk probably sends 183/200 back to callee, finalizing this codec 
negotiation. VP8 is chosen for this call leg.
  *   Asterisk will INVITE Called
  *   Called sends '200 OK' to Asterisk requesting H264
  * yes
  *   Asterisk Sends '200 OK' back to Kamailio to be relayed to Caller 
accepting VP8
  * No. I assume that the callee didn't accept VP8 because it doesn't 
support it. It instead chose the highest priority codec it supported among the 
codecs that Asterisk advertised.
  * Asterisk accepts this codec because it supports it. There is no test 
checking which codec was chosen on the caller side and there is no 
renegotiation. Hence, video is broken forever.

Again, if you could show us the details (like attaching the SDPs that I asked 
for previously), then we wouldn't have to guess what's happening. We could tell 
you for sure.


Med vennlig hilsen
Pan B. Christensen

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 22:27
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kama

Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Wilkins, Steve]

Thank you Pan. I am going to continue to look at this this weekend.  I'm not 
sure why Asterisk is only offering 99.
I'll take a look at it with fresh eyes today, although I've been on this for 
two weeks now.

It is strange that the call works if I by-pass Kamailio;  Kamailio is just 
relaying (I think?).

Thank you SO much again Pan,
Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Saturday, June 9, 2018 6:20 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

I have to admit that I don't have much experience with video calls, but as far 
as I can tell VP8 is negotiated with WebRTC client. Nothing is negotiated with 
softphone.

WebRTC client offers:
Media Description, name and address (m): video 55185 UDP/TLS/RTP/SAVPF 96 97 98 
99 100 101 102 123 127 122 125 107 108 109 124

Asterisk offers to softphone:
Media Description, name and address (m): video 15112 RTP/AVP 99
Why is it advertising only 99?

Softphone answers with:
Media Description, name and address (m): video 36140 RTP/AVP 115 113 34 31
No match with offer, video is not negotiated for this call leg.

Asterisk answers WebRTC with:
Media Description, name and address (m): video 10328 UDP/TLS/RTP/SAVPF 96 100
96 is chosen for this call leg.


With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: lørdag 9. juni 2018 03:59
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello again Pan and Thank You for looking at this!!  Here is the pcap for the 
call that is not showing video on either side of the call.  As I mentioned I do 
get two way Audio, the call stays connected, but just no Video.  The Video 
ports look to be correct.  Video packets appear to go out (at least packets are 
going out over the Video ports). The screen on the Caller (A WebRTC Client) and 
the Called (A Provider Phone) are both blank.  The WebRTC client is Registered 
in Kamailio

Note: when I sanitized the file using tcprewrite an extra RTP packet appeared 
(Unknown RTP packet version 0), this was not here in the original.

I use Kamailio 5.2.0 and Asterisk 15.3

Thank you again!,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 5:12 PM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Dear Steve.

Asterisk is a B2BUA. It doesn't just route SIP messages like Kamailio does. 
Most of the time, it answers the call, sets up a new call to the other party 
and then bridges the two calls, making two separate call IDs (depending on your 
configuration).

Additional comments in blue inline below:


  *   INVITE from Caller to Kamailio (this is where the Caller is Registered) 
offering VP8
  *   Kamailio will relay the INVITE to Asterisk
  * Asterisk probably sends 183/200 back to callee, finalizing this codec 
negotiation. VP8 is chosen for this call leg.
  *   Asterisk will INVITE Called
  *   Called sends '200 OK' to Asterisk requesting H264
  * yes
  *   Asterisk Sends '200 OK' back to Kamailio to be relayed to Caller 
accepting VP8
  * No. I assume that the callee didn't accept VP8 because it doesn't 
support it. It instead chose the highest priority codec it supported among the 
codecs that Asterisk advertised.
  * Asterisk accepts this codec because it supports it. There is no test 
checking which codec was chosen on the caller side and there is no 
renegotiation. Hence, video is broken forever.

Again, if you could show us the details (like attaching the SDPs that I asked 
for previously), then we wouldn't have to guess what's happening. We could tell 
you for sure.


Med vennlig hilsen
Pan B. Christensen

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 22:27
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi Pan,

If the Registered caller is a WebRTC client (Caller) whose preferred coded is 
VP8, calls a Phone (Called) whose preferred Video is H264, I should see


  *   INVITE from Caller to Kamailio (this is where the Caller is Registered) 
offering VP8
  *   Kamailio will relay the INVITE to Asterisk
  *   Asterisk will INVITE Called
  *   Called sends '200 OK' to Asterisk requesting H264
  *   Asterisk Sends '200 OK' back to Kamailio to be relayed to Caller 
accepting VP8

Is this correct?  Then I think one of the issues is that there is no fmtp line 
in the VP8.  The only codecs that have fmtp lines is for the H264 codecs.

Thank you,
-Steve

F

Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Wilkins, Steve]

Sorry Pan, but one other thing I noticed.  When Asterisk send the 200 OK, 
Kamailio does see it, but I don't see where it is relaying it to the WebRTC 
client that made the call.  However, it must, otherwise the call would 
disconnect in 30 seconds and also, I do have two-way Audio.

Thanks again for you input on this. It's seems a bit confusing as to where an 
issue lies.

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 11:06 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

That's not strange. The soft phone probably doesn't advertise support for VP8, 
so it's not chosen.
Calling from WebRTC, I assume that VP8 is the preferred codec and Asterisk 
accepts it. Asterisk then finds out that the soft phone doesn't support VP8 and 
negotiates a different codec with that client.

If you show us the SDPs of INVITE and 183/200 on both sides of Asterisk (in the 
order they are sent), we can tell you exactly what happens. Failing that, I'd 
say that the culprit is Asterisk, which probably negotiates two different 
codecs without the ability to transcode.

With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 15:09
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Yes, for example a WebRTC Client (VP8) calls a Soft-Phone (H264).  What is 
strange is that if it is the other way around and the Soft-Phone calls the 
WebRTC client, it works.

Thank you

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 9:00 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello Steve.

Does Asterisk negotiate different codecs with each client? If so, it needs to 
transcode, which I believe is currently not supported for video. What does 
Asterisk send back to device A?

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 14:03
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi All,

Issue: when a Call is made through Kamailio and Asterisk.  Asterisk uses 
incorrect Video RTP Payload Type when sending Video packets.

I have a situation where I make a call from Device A to Device B and Device A 
is Registered in Kamailio.  When Device A Calls Device B, Kamailio sends an 
'INVITE' to Asterisk, Asterisk then  'INVITES' Device B.  I get two-way Audio, 
the call stays connected, however, when Video packets are sent to Device B, the 
RTP Payload Type is incorrect.  The port is correct, but just not the Payload 
Type.

Here is where I think Kamailio is involved. In the first Invite from Kamailio 
to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, 
Device B wants to use '115 H264' and when Asterisk sends out Video packets, it 
is using '100' instead of '115', and of course I have no Video.  I don't know 
if this is just a coincidence but it sure seems like that is where the issue 
may lie.

Has anyone ever seen this behavior?  The Asterisk teams does not think it's an 
Asterisk issue.

Thank you,
-Steve
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Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Wilkins, Steve]

I forgot to mention that if I by-pass Kamailio, the call works.  I would like 
to send the pcap, but it has IP-Address that can't be shared.

Thank you!

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 11:06 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

That's not strange. The soft phone probably doesn't advertise support for VP8, 
so it's not chosen.
Calling from WebRTC, I assume that VP8 is the preferred codec and Asterisk 
accepts it. Asterisk then finds out that the soft phone doesn't support VP8 and 
negotiates a different codec with that client.

If you show us the SDPs of INVITE and 183/200 on both sides of Asterisk (in the 
order they are sent), we can tell you exactly what happens. Failing that, I'd 
say that the culprit is Asterisk, which probably negotiates two different 
codecs without the ability to transcode.

With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 15:09
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Yes, for example a WebRTC Client (VP8) calls a Soft-Phone (H264).  What is 
strange is that if it is the other way around and the Soft-Phone calls the 
WebRTC client, it works.

Thank you

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 9:00 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello Steve.

Does Asterisk negotiate different codecs with each client? If so, it needs to 
transcode, which I believe is currently not supported for video. What does 
Asterisk send back to device A?

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 14:03
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi All,

Issue: when a Call is made through Kamailio and Asterisk.  Asterisk uses 
incorrect Video RTP Payload Type when sending Video packets.

I have a situation where I make a call from Device A to Device B and Device A 
is Registered in Kamailio.  When Device A Calls Device B, Kamailio sends an 
'INVITE' to Asterisk, Asterisk then  'INVITES' Device B.  I get two-way Audio, 
the call stays connected, however, when Video packets are sent to Device B, the 
RTP Payload Type is incorrect.  The port is correct, but just not the Payload 
Type.

Here is where I think Kamailio is involved. In the first Invite from Kamailio 
to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, 
Device B wants to use '115 H264' and when Asterisk sends out Video packets, it 
is using '100' instead of '115', and of course I have no Video.  I don't know 
if this is just a coincidence but it sure seems like that is where the issue 
may lie.

Has anyone ever seen this behavior?  The Asterisk teams does not think it's an 
Asterisk issue.

Thank you,
-Steve
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Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Wilkins, Steve]

Hi Pan,

If the Registered caller is a WebRTC client (Caller) whose preferred coded is 
VP8, calls a Phone (Called) whose preferred Video is H264, I should see


  *   INVITE from Caller to Kamailio (this is where the Caller is Registered) 
offering VP8
  *   Kamailio will relay the INVITE to Asterisk
  *   Asterisk will INVITE Called
  *   Called sends '200 OK' to Asterisk requesting H264
  *   Asterisk Sends '200 OK' back to Kamailio to be relayed to Caller 
accepting VP8

Is this correct?  Then I think one of the issues is that there is no fmtp line 
in the VP8.  The only codecs that have fmtp lines is for the H264 codecs.

Thank you,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 11:06 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

That's not strange. The soft phone probably doesn't advertise support for VP8, 
so it's not chosen.
Calling from WebRTC, I assume that VP8 is the preferred codec and Asterisk 
accepts it. Asterisk then finds out that the soft phone doesn't support VP8 and 
negotiates a different codec with that client.

If you show us the SDPs of INVITE and 183/200 on both sides of Asterisk (in the 
order they are sent), we can tell you exactly what happens. Failing that, I'd 
say that the culprit is Asterisk, which probably negotiates two different 
codecs without the ability to transcode.

With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 15:09
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Yes, for example a WebRTC Client (VP8) calls a Soft-Phone (H264).  What is 
strange is that if it is the other way around and the Soft-Phone calls the 
WebRTC client, it works.

Thank you

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 9:00 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello Steve.

Does Asterisk negotiate different codecs with each client? If so, it needs to 
transcode, which I believe is currently not supported for video. What does 
Asterisk send back to device A?

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 14:03
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi All,

Issue: when a Call is made through Kamailio and Asterisk.  Asterisk uses 
incorrect Video RTP Payload Type when sending Video packets.

I have a situation where I make a call from Device A to Device B and Device A 
is Registered in Kamailio.  When Device A Calls Device B, Kamailio sends an 
'INVITE' to Asterisk, Asterisk then  'INVITES' Device B.  I get two-way Audio, 
the call stays connected, however, when Video packets are sent to Device B, the 
RTP Payload Type is incorrect.  The port is correct, but just not the Payload 
Type.

Here is where I think Kamailio is involved. In the first Invite from Kamailio 
to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, 
Device B wants to use '115 H264' and when Asterisk sends out Video packets, it 
is using '100' instead of '115', and of course I have no Video.  I don't know 
if this is just a coincidence but it sure seems like that is where the issue 
may lie.

Has anyone ever seen this behavior?  The Asterisk teams does not think it's an 
Asterisk issue.

Thank you,
-Steve
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Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Wilkins, Steve]

Yes, for example a WebRTC Client (VP8) calls a Soft-Phone (H264).  What is 
strange is that if it is the other way around and the Soft-Phone calls the 
WebRTC client, it works.

Thank you

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 9:00 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello Steve.

Does Asterisk negotiate different codecs with each client? If so, it needs to 
transcode, which I believe is currently not supported for video. What does 
Asterisk send back to device A?

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 14:03
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi All,

Issue: when a Call is made through Kamailio and Asterisk.  Asterisk uses 
incorrect Video RTP Payload Type when sending Video packets.

I have a situation where I make a call from Device A to Device B and Device A 
is Registered in Kamailio.  When Device A Calls Device B, Kamailio sends an 
'INVITE' to Asterisk, Asterisk then  'INVITES' Device B.  I get two-way Audio, 
the call stays connected, however, when Video packets are sent to Device B, the 
RTP Payload Type is incorrect.  The port is correct, but just not the Payload 
Type.

Here is where I think Kamailio is involved. In the first Invite from Kamailio 
to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, 
Device B wants to use '115 H264' and when Asterisk sends out Video packets, it 
is using '100' instead of '115', and of course I have no Video.  I don't know 
if this is just a coincidence but it sure seems like that is where the issue 
may lie.

Has anyone ever seen this behavior?  The Asterisk teams does not think it's an 
Asterisk issue.

Thank you,
-Steve
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https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Wilkins, Steve]

Hi All,

Issue: when a Call is made through Kamailio and Asterisk.  Asterisk uses 
incorrect Video RTP Payload Type when sending Video packets.

I have a situation where I make a call from Device A to Device B and Device A 
is Registered in Kamailio.  When Device A Calls Device B, Kamailio sends an 
'INVITE' to Asterisk, Asterisk then  'INVITES' Device B.  I get two-way Audio, 
the call stays connected, however, when Video packets are sent to Device B, the 
RTP Payload Type is incorrect.  The port is correct, but just not the Payload 
Type.

Here is where I think Kamailio is involved. In the first Invite from Kamailio 
to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, 
Device B wants to use '115 H264' and when Asterisk sends out Video packets, it 
is using '100' instead of '115', and of course I have no Video.  I don't know 
if this is just a coincidence but it sure seems like that is where the issue 
may lie.

Has anyone ever seen this behavior?  The Asterisk teams does not think it's an 
Asterisk issue.

Thank you,
-Steve
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[SR-Users] Thank You

[Wilkins, Steve]

Hi All,

I normally get on this board to ask questions but today I just wanted to take a 
minute and say -

Thank you to all of you who have taken the time to help me.  Some of you have 
spent a lot of time schooling me
on Kamailio, its Interactions with a PBX and different clients.  I have learned 
tons from this board and just
want to say again, Thank You All!

-Steve

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[SR-Users] General Kamailio with Asterisk (or another PBX) Opinion

[Wilkins, Steve]

HI All,

I am curious, for those of you who use Kamailio with Asterisk or another PBX.
Who forwards Registrations to Asterisk or PBX, and who lets Kamailio maintain 
Registrations?

Thanks All!
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Re: [SR-Users] Switching form UDP to TCP causes authentication errors

[Wilkins, Steve]

Thank you for your response.  

I do not have this issue with UDP, it is when I switch to TCP that an INVITE is 
sending the Unauthorized.

If I attempt something like , the code below, the INVITE is not sent out again 
(as I thought it would be).  I see the ACK being relayed after the 
Authentication error, and that is it.

if (t_check_status("401")) {
if (!auth_check("$fd", "sipusers", "1")) {
  auth_challenge("$fd", "0");
  exit;
}
}  

I did attempt the uac_auth() as you suggested, but I got no further.

Thanks again!

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Daniel 
Tryba
Sent: Friday, May 25, 2018 9:47 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Switching form UDP to TCP causes authentication errors

On Thu, May 24, 2018 at 11:29:54PM +, Wilkins, Steve wrote:
> When I switched from UDP to TCP I started getting Authentication 
> Errors, Asterisk responds to an INVITE via Kamailio with a '401 
> Unauthorized', but Kamailio does nothing with it.

Did you get 401 responses using UDP? Do you have any logic in place to respond 
to the 401? By default kamailio does nothing. 

> Processing just stops near WITH_BLOCK401407.  Shouldn't the 401 be 
> relayed so a new INVITE can be sent?

What does "stops near" mean? What does WITH_BLOCK401407 do? The name suggest to 
block/do nothing with 401 and 407s.

What you need to do is call uac_auth() with the correct credentials to respond 
to the challenge:
https://www.kamailio.org/docs/modules/stable/modules/uac.html#uac.f.uac_auth

Or is the problem Asterisk challenges INVITEs using TCP? 

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[SR-Users] Switching form UDP to TCP causes authentication errors

[Wilkins, Steve]

Hello All,

When I switched from UDP to TCP I started getting Authentication Errors, 
Asterisk responds to an INVITE via Kamailio with a '401 Unauthorized', but 
Kamailio does nothing with it.  Processing just stops near WITH_BLOCK401407.  
Shouldn't the 401 be relayed so a new INVITE can be sent?

Thank you.
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Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK sent to it by Asterisk

[Wilkins, Steve]

Got it!  Thank you Federico!

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Federico Cabiddu
Sent: Tuesday, May 22, 2018 8:09 AM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK 
sent to it by Asterisk

You have to  listen on two different ports and advertise them according to you 
routing if you have only one interface (IP) where your Kamailio is listening to.

Cheers,

Federico

On Tue, May 22, 2018 at 2:02 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
Good morning and thank you for your response.

I thought that advertise took care of listening on Public and Private IP 
Address.

I did try -
listen=udp:Private-IP-Address:5060 advertise 
34.226.187.61:5060<http://34.226.187.61:5060>
listen=udp:Public-IP-Address:5060
but I got errors without changing the port, My Asterisk only communicate on 
5060, so changing the port would not help me.

I do have modparam("rr", "enable_double_rr", 2); I was not sure if mhomed 
needed to be set in this scenario.

I have done a lot of searching and have not many examples of the situation I am 
facing.

I think I am just missing something.

Thanks again!

.



From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Tuesday, May 22, 2018 2:35 AM

To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK 
sent to it by Asterisk

Good morning,
so, if I understand correctly Kamailio is listening on a single interface which 
is natted.
You need two R-R, one with the public and one with the private IP. You also 
need to put Kamailio on listen on two different ports and use the listen 
directive along with its advertise parameter.

Regards,

Federico

On Tue, May 22, 2018 at 12:12 AM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
What is actually happening is –

I M USING Kamailio 5.2 with Asterisk 14.6 and 15.3.  Asterisk cannot reach 
Kamailio on Public IP (Per design)

Asterisk 14.6: If Asterisk sends an ‘INVITE’ to Kamailio (on Private IP), then 
Kamailio sends back a ‘200 OK’ with its Public IP in both Record-Routes.
Asterisk 14.6 is changing the first Record-Route to the Private-IP of Kamailio 
and also changes the Via to The same Private IP, and hence the ACK is
sent to and received by Kamailio…Call works perfect. (Asterisk developers say 
they are not doing this!).

Asterisk >= 15.0: The above mentioned Record-Route and Via do not get changed 
and hence the ACK is sent to Kamailio’ s Public IP, but never makes it there 
(because the my firewall will not allow it).

This is why I was wondering if I could change the Record-Route on the ‘200  OK’ 
that is sent to Asterisk.  I just can’t figure out how to do it.

Thank you for your time!

From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Monday, May 21, 2018 5:06 PM

To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK 
sent to it by Asterisk

Not sure I understand your scenario.
How many interfaces your kamailio has?
How's the callflow going?
Public->Kamailio->Asterisk?
Did you have a look on "advertise" parameter of core "listen" directive?
https://www.kamailio.org/wiki/cookbooks/5.1.x/core#listen

Best regards,

Federico

On Mon, May 21, 2018 at 7:32 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
Hello,

I set that double route to ‘2’ and Kamailio still not hearing traffic coming in 
from Public.  I think I need to figure out how to get Kamailio to set the first 
RR in the ‘200 OK’ to its Private IP, because  the Server Kamailio is on may 
not be open to public traffic.

Thank you

From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Monday, May 21, 2018 1:08 PM
To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK 
sent to it by Asterisk

Hi,
if you have Kamailio listening on private IP and public IP you need two 
record-routes.
Have a look at this param of the rr module:
http://www.kamailio.org/docs/modules/devel/modules/rr.html#rr.p.enable_double_rr

Best regards,

Federico


On Mon, May 21, 2018 at 7:01 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:

Hello All,

When Kama

Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK sent to it by Asterisk

[Wilkins, Steve]

Good morning and thank you for your response.

I thought that advertise took care of listening on Public and Private IP 
Address.

I did try -
listen=udp:Private-IP-Address:5060 advertise 34.226.187.61:5060
listen=udp:Public-IP-Address:5060
but I got errors without changing the port, My Asterisk only communicate on 
5060, so changing the port would not help me.

I do have modparam("rr", "enable_double_rr", 2); I was not sure if mhomed 
needed to be set in this scenario.

I have done a lot of searching and have not many examples of the situation I am 
facing.

I think I am just missing something.

Thanks again!

.



From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Federico Cabiddu
Sent: Tuesday, May 22, 2018 2:35 AM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK 
sent to it by Asterisk

Good morning,
so, if I understand correctly Kamailio is listening on a single interface which 
is natted.
You need two R-R, one with the public and one with the private IP. You also 
need to put Kamailio on listen on two different ports and use the listen 
directive along with its advertise parameter.

Regards,

Federico

On Tue, May 22, 2018 at 12:12 AM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
What is actually happening is –

I M USING Kamailio 5.2 with Asterisk 14.6 and 15.3.  Asterisk cannot reach 
Kamailio on Public IP (Per design)

Asterisk 14.6: If Asterisk sends an ‘INVITE’ to Kamailio (on Private IP), then 
Kamailio sends back a ‘200 OK’ with its Public IP in both Record-Routes.
Asterisk 14.6 is changing the first Record-Route to the Private-IP of Kamailio 
and also changes the Via to The same Private IP, and hence the ACK is
sent to and received by Kamailio…Call works perfect. (Asterisk developers say 
they are not doing this!).

Asterisk >= 15.0: The above mentioned Record-Route and Via do not get changed 
and hence the ACK is sent to Kamailio’ s Public IP, but never makes it there 
(because the my firewall will not allow it).

This is why I was wondering if I could change the Record-Route on the ‘200  OK’ 
that is sent to Asterisk.  I just can’t figure out how to do it.

Thank you for your time!

From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Monday, May 21, 2018 5:06 PM

To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK 
sent to it by Asterisk

Not sure I understand your scenario.
How many interfaces your kamailio has?
How's the callflow going?
Public->Kamailio->Asterisk?
Did you have a look on "advertise" parameter of core "listen" directive?
https://www.kamailio.org/wiki/cookbooks/5.1.x/core#listen

Best regards,

Federico

On Mon, May 21, 2018 at 7:32 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
Hello,

I set that double route to ‘2’ and Kamailio still not hearing traffic coming in 
from Public.  I think I need to figure out how to get Kamailio to set the first 
RR in the ‘200 OK’ to its Private IP, because  the Server Kamailio is on may 
not be open to public traffic.

Thank you

From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Monday, May 21, 2018 1:08 PM
To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK 
sent to it by Asterisk

Hi,
if you have Kamailio listening on private IP and public IP you need two 
record-routes.
Have a look at this param of the rr module:
http://www.kamailio.org/docs/modules/devel/modules/rr.html#rr.p.enable_double_rr

Best regards,

Federico


On Mon, May 21, 2018 at 7:01 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:

Hello All,

When Kamailio  sends a ‘200 OK’ to Asterisk, it is putting its Public IP into 
the Record-Routes.  In Asterisk 14.6, it would send the ‘ACK’ back to the 
Private IP Address of Kamailio, but Asterisk 15.x is using the Public IP 
Address that Kamailio placed in the Record-Routes so…
Is there a way to force Kamailio to set the Record-Route to its Private IP 
address of the first Record-Route in messages forwarded to Asterisk?

Thank you?

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Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK sent to it by Asterisk

[Wilkins, Steve]

What is actually happening is –

I M USING Kamailio 5.2 with Asterisk 14.6 and 15.3.  Asterisk cannot reach 
Kamailio on Public IP (Per design)

Asterisk 14.6: If Asterisk sends an ‘INVITE’ to Kamailio (on Private IP), then 
Kamailio sends back a ‘200 OK’ with its Public IP in both Record-Routes.
Asterisk 14.6 is changing the first Record-Route to the Private-IP of Kamailio 
and also changes the Via to The same Private IP, and hence the ACK is
sent to and received by Kamailio…Call works perfect. (Asterisk developers say 
they are not doing this!).

Asterisk >= 15.0: The above mentioned Record-Route and Via do not get changed 
and hence the ACK is sent to Kamailio’ s Public IP, but never makes it there 
(because the my firewall will not allow it).

This is why I was wondering if I could change the Record-Route on the ‘200  OK’ 
that is sent to Asterisk.  I just can’t figure out how to do it.

Thank you for your time!

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Federico Cabiddu
Sent: Monday, May 21, 2018 5:06 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK 
sent to it by Asterisk

Not sure I understand your scenario.
How many interfaces your kamailio has?
How's the callflow going?
Public->Kamailio->Asterisk?
Did you have a look on "advertise" parameter of core "listen" directive?
https://www.kamailio.org/wiki/cookbooks/5.1.x/core#listen

Best regards,

Federico

On Mon, May 21, 2018 at 7:32 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
Hello,

I set that double route to ‘2’ and Kamailio still not hearing traffic coming in 
from Public.  I think I need to figure out how to get Kamailio to set the first 
RR in the ‘200 OK’ to its Private IP, because  the Server Kamailio is on may 
not be open to public traffic.

Thank you

From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Monday, May 21, 2018 1:08 PM
To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK 
sent to it by Asterisk

Hi,
if you have Kamailio listening on private IP and public IP you need two 
record-routes.
Have a look at this param of the rr module:
http://www.kamailio.org/docs/modules/devel/modules/rr.html#rr.p.enable_double_rr

Best regards,

Federico


On Mon, May 21, 2018 at 7:01 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:

Hello All,

When Kamailio  sends a ‘200 OK’ to Asterisk, it is putting its Public IP into 
the Record-Routes.  In Asterisk 14.6, it would send the ‘ACK’ back to the 
Private IP Address of Kamailio, but Asterisk 15.x is using the Public IP 
Address that Kamailio placed in the Record-Routes so…
Is there a way to force Kamailio to set the Record-Route to its Private IP 
address of the first Record-Route in messages forwarded to Asterisk?

Thank you?

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Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK sent to it by Asterisk

[Wilkins, Steve]

Hello,

I set that double route to ‘2’ and Kamailio still not hearing traffic coming in 
from Public.  I think I need to figure out how to get Kamailio to set the first 
RR in the ‘200 OK’ to its Private IP, because  the Server Kamailio is on may 
not be open to public traffic.

Thank you

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Federico Cabiddu
Sent: Monday, May 21, 2018 1:08 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] I think I found issue to Kamailio not forwarding ACK 
sent to it by Asterisk

Hi,
if you have Kamailio listening on private IP and public IP you need two 
record-routes.
Have a look at this param of the rr module:
http://www.kamailio.org/docs/modules/devel/modules/rr.html#rr.p.enable_double_rr

Best regards,

Federico


On Mon, May 21, 2018 at 7:01 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:

Hello All,

When Kamailio  sends a ‘200 OK’ to Asterisk, it is putting its Public IP into 
the Record-Routes.  In Asterisk 14.6, it would send the ‘ACK’ back to the 
Private IP Address of Kamailio, but Asterisk 15.x is using the Public IP 
Address that Kamailio placed in the Record-Routes so…
Is there a way to force Kamailio to set the Record-Route to its Private IP 
address of the first Record-Route in messages forwarded to Asterisk?

Thank you?

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[SR-Users] I think I found issue to Kamailio not forwarding ACK sent to it by Asterisk

[Wilkins, Steve]

Hello All,

When Kamailio  sends a '200 OK' to Asterisk, it is putting its Public IP into 
the Record-Routes.  In Asterisk 14.6, it would send the 'ACK' back to the 
Private IP Address of Kamailio, but Asterisk 15.x is using the Public IP 
Address that Kamailio placed in the Record-Routes so...
Is there a way to force Kamailio to set the Record-Route to its Private IP 
address of the first Record-Route in messages forwarded to Asterisk?

Thank you?
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[SR-Users] Cannot get rid of following errors, and wondering if this is causing some of my issues

[Wilkins, Steve]

ERROR:  [core/pvapi.c:1452]: pv_printf(): no more space for spec 
value
ERROR:  [core/pvapi.c:1461]: pv_printf(): buffer overflow -- increase the 
buffer size...

I have
pv_buffer_size=16384;
tcp_rd_buf_size=16384;

Any ideas?

Thank you,
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[SR-Users] Supper confused on Kamailio with Asterisk Call

[Wilkins, Steve]

Hi All,

I am using Kamailio 5.2 and Asterisk (14.6 & 15.x) and I am having a very 
strange issue.  If I use Asterisk 14.6, the call (WebRTC<=>WebRTC) works 
perfectly.  However, if I use Asterisk 15.x, the call drops in 30 seconds.  In 
comparing tcpdump files, the first place I see a difference is in the '200 OK' 
response to Kamailio' s 'INVITE to Asterisk'; in Asterisk 15.3 the first record 
Record-Route is Kamailio Public IP Address, however, in the Asterisk 14.6 (the 
working call) it is Kamailio' s Private IP Address.  Asterisk Developers told 
me it is Kamailio doing this...and I am confused.  Is this true, and if so why 
would it be different based on the Asterisk version?

Thank you!
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[SR-Users] WebRTC to WebRTC call Kamailio ACK Problem

[Wilkins, Steve]

Hello,

I am  using Kamailio 5.2 and Asterisk 15.3 and while doing a WebRTC (client1) 
to WebRTC (client2) call, Asterisks sends a re-invite and Kamailio responds 
with a '200 OK', but when Asterisk sends back the "ACK" to Kamailio, it cannot 
be relayed.

Errors =>
WARNING:  [core/msg_translator.c:2767]: via_builder(): TCP/TLS connection 
(id: 0) for WebSocket could not be found
ERROR:  [core/msg_translator.c:1982]: build_req_buf_from_sip_req(): could 
not create Via header
ERROR:  [core/forward.c:549]: forward_request(): building failed
ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm 
terribly sorry, server error occurred (1/SL)

As a side note, I still occasionally see the Errors =>
ERROR:  [core/pvapi.c:1452]: pv_printf(): no more space for spec value
ERROR:  [core/pvapi.c:1461]: pv_printf(): buffer overflow -- increase the 
buffer size...
No matter how large I make pv_buffer_size, and tcp_rd_buf_size;   
pv_buffer_size=16384, and tcp_rd_buf_size=16384;

Thank you all!


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Re: [SR-Users] Transport issue thought

[Wilkins, Steve]

After some more debugging I see that (for a WebRTC<=>WebRTC call) if using 
Kamailio 5.2 (UDP) and Asterisk 15.x, a third INVITE occurs from Asterisk back 
to Kamailio.  Kamailio is unable to relay because it thinks the web socket is 
down.  There error is =>
WARNING:  [core/msg_translator.c:2767]: via_builder(): TCP/TLS connection 
(id: 0) for WebSocket could not be found
ERROR:  [core/msg_translator.c:1982]: build_req_buf_from_sip_req(): could 
not create Via header

However, if using Kamailio 5.2 and Asterisk 14.6, a third INVITE does not occur 
and everything works perfect!

I don’t know if this is a Kamailio or Asterisk issue.  I have worked on this 
one a long time and I sure hope I’m close 

Thank you everyone!

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Wilkins, Steve
Sent: Friday, May 18, 2018 3:58 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] Transport issue thought

I think the reason for the second issue listed below is that Kamailio never 
receives the ‘200 OK’ from back from the INVITE to Asterisk.  I have noticed 
that starting sometime after version 14.6, Asterisk puts its Public IP-Address 
in the Contact field where as in 14.6 it puts it Private IP-Address.

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Wilkins, Steve
Sent: Friday, May 18, 2018 12:15 PM
To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Transport issue thought

I sent the log but I just wanted to clear up the issue.

This is a WebRTC to WebRTC call

  *   If I use Kamailio 5.2 (UDP) and Asterisk 14.6, the call works perfect!
  *   If I use Kamailio 5.2 (UDP) and Asterisk 15.3, the call connects but 
disconnects after 30 seconds (Asterisk never received ACK from ‘200 OK’)
  *   If I use Kamailio 5.2 (TCP) and Asterisk x.x the call never connects.


Thank you!



From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Federico Cabiddu
Sent: Friday, May 18, 2018 11:02 AM
To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Transport issue thought

I don't understand, are you able to send an INVITE to Asterisk via TCP?
Could you share a trace of the calls, masking the sensible information?

Cheers,

Federico

On Fri, May 18, 2018 at 4:56 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
I have tried using t_relay_to_tcp() and t_realy, both with and without setting 
$ru = $ru + ";transport=tcp" right before the call.
I get different errors and failures depending on which way I make the call.  
Which log would you want to see?


  *   I also wanted to point out that if I use UDP, the call connects (it is a 
WebRTC client to WebRTC client call, and both clients are registered in 
Kamailio).  The call just gets dropped after 30 seconds because Asterisk 15.3 
is not receiving the ACK back to the ‘200 OK’.  Aas stated, TCP does not even 
connect.


  *   Another very interesting thing is that if I use UDP and Asterisk 14.6, 
the call works perfect! It stays connected and I have full duplex Audio and 
Video

Thank you!


From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Friday, May 18, 2018 8:49 AM

To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Transport issue thought

Can you print the logs when it tries to send the ACK?
Check also that the ACK ruri contains the same destination port which was used 
for the INVITE, otherwise a new connection will be created.

Best regards,

Federico

On Fri, May 18, 2018 at 2:31 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
Thank you Alex and Federico,

I verified, and SO_REUSEPORT is defined on my OS.  I am using Kamailio 5.2 and 
I set ‘tcp_reuse_port=yes;’ and $fs;  this has been to no avail as ‘ACK’s’ are 
still using the high port number randomly assigned by Kamailio.

Thank you all for sharing your knowledge!


From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Friday, May 18, 2018 12:57 AM
To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Transport issue thought

You are right Alex, Linux kernel didn't support SO_REUSEPORT, which allows a 
socket to be used as source for a tcp connection while is already bound, until 
version 3.9.
Kamailio's parameter tcp_reuse_port, if enabled and your OS has support for 
SO_REUSEPORT (so not only Linux but FreeBSD, OSX and others), allows you to use 
force_send_s

Re: [SR-Users] Transport issue thought

[Wilkins, Steve]

I think the reason for the second issue listed below is that Kamailio never 
receives the ‘200 OK’ from back from the INVITE to Asterisk.  I have noticed 
that starting sometime after version 14.6, Asterisk puts its Public IP-Address 
in the Contact field where as in 14.6 it puts it Private IP-Address.

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Wilkins, Steve
Sent: Friday, May 18, 2018 12:15 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] Transport issue thought

I sent the log but I just wanted to clear up the issue.

This is a WebRTC to WebRTC call

  *   If I use Kamailio 5.2 (UDP) and Asterisk 14.6, the call works perfect!
  *   If I use Kamailio 5.2 (UDP) and Asterisk 15.3, the call connects but 
disconnects after 30 seconds (Asterisk never received ACK from ‘200 OK’)
  *   If I use Kamailio 5.2 (TCP) and Asterisk x.x the call never connects.


Thank you!



From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Federico Cabiddu
Sent: Friday, May 18, 2018 11:02 AM
To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Transport issue thought

I don't understand, are you able to send an INVITE to Asterisk via TCP?
Could you share a trace of the calls, masking the sensible information?

Cheers,

Federico

On Fri, May 18, 2018 at 4:56 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
I have tried using t_relay_to_tcp() and t_realy, both with and without setting 
$ru = $ru + ";transport=tcp" right before the call.
I get different errors and failures depending on which way I make the call.  
Which log would you want to see?


  *   I also wanted to point out that if I use UDP, the call connects (it is a 
WebRTC client to WebRTC client call, and both clients are registered in 
Kamailio).  The call just gets dropped after 30 seconds because Asterisk 15.3 
is not receiving the ACK back to the ‘200 OK’.  Aas stated, TCP does not even 
connect.


  *   Another very interesting thing is that if I use UDP and Asterisk 14.6, 
the call works perfect! It stays connected and I have full duplex Audio and 
Video

Thank you!


From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Friday, May 18, 2018 8:49 AM

To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Transport issue thought

Can you print the logs when it tries to send the ACK?
Check also that the ACK ruri contains the same destination port which was used 
for the INVITE, otherwise a new connection will be created.

Best regards,

Federico

On Fri, May 18, 2018 at 2:31 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
Thank you Alex and Federico,

I verified, and SO_REUSEPORT is defined on my OS.  I am using Kamailio 5.2 and 
I set ‘tcp_reuse_port=yes;’ and $fs;  this has been to no avail as ‘ACK’s’ are 
still using the high port number randomly assigned by Kamailio.

Thank you all for sharing your knowledge!


From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Friday, May 18, 2018 12:57 AM
To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Transport issue thought

You are right Alex, Linux kernel didn't support SO_REUSEPORT, which allows a 
socket to be used as source for a tcp connection while is already bound, until 
version 3.9.
Kamailio's parameter tcp_reuse_port, if enabled and your OS has support for 
SO_REUSEPORT (so not only Linux but FreeBSD, OSX and others), allows you to use 
force_send_socket (or $fs) to send messages from a TCP port kamailio is 
listening to.

Cheers,

Federico

On Fri, May 18, 2018 at 12:28 AM, Alex Balashov 
<abalas...@evaristesys.com<mailto:abalas...@evaristesys.com>> wrote:
When an outgoing TCP connection is opened, either a port can be
explicitly bound, or it is auto-assigned by the OS's networking stack. I
believe Kamailio does the latter and does not offer options to constrain
the range. If it does, I'm not aware of any apart from this one:

https://www.kamailio.org/wiki/cookbooks/5.1.x/core#tcp_reuse_port

Not sure if it would help in this case, you'd have to give it a try.

On Thu, May 17, 2018 at 10:21:34PM +, Wilkins, Steve wrote:

> It appears that the bottom line of my TCP transport not working is that 
> Kamailio is randomly assigning large port numbers to send the TCP traffic out 
> on.
> I am not able to randomly open high ports for this purpose.  Is there a way 
> to tell Kamailio to only use specific ports for this.  I have tried u

Re: [SR-Users] Transport issue thought

[Wilkins, Steve]

I sent the log but I just wanted to clear up the issue.

This is a WebRTC to WebRTC call

  *   If I use Kamailio 5.2 (UDP) and Asterisk 14.6, the call works perfect!
  *   If I use Kamailio 5.2 (UDP) and Asterisk 15.3, the call connects but 
disconnects after 30 seconds (Asterisk never received ACK from ‘200 OK’)
  *   If I use Kamailio 5.2 (TCP) and Asterisk x.x the call never connects.


Thank you!



From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Federico Cabiddu
Sent: Friday, May 18, 2018 11:02 AM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] Transport issue thought

I don't understand, are you able to send an INVITE to Asterisk via TCP?
Could you share a trace of the calls, masking the sensible information?

Cheers,

Federico

On Fri, May 18, 2018 at 4:56 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
I have tried using t_relay_to_tcp() and t_realy, both with and without setting 
$ru = $ru + ";transport=tcp" right before the call.
I get different errors and failures depending on which way I make the call.  
Which log would you want to see?


  *   I also wanted to point out that if I use UDP, the call connects (it is a 
WebRTC client to WebRTC client call, and both clients are registered in 
Kamailio).  The call just gets dropped after 30 seconds because Asterisk 15.3 
is not receiving the ACK back to the ‘200 OK’.  Aas stated, TCP does not even 
connect.


  *   Another very interesting thing is that if I use UDP and Asterisk 14.6, 
the call works perfect! It stays connected and I have full duplex Audio and 
Video

Thank you!


From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Friday, May 18, 2018 8:49 AM

To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Transport issue thought

Can you print the logs when it tries to send the ACK?
Check also that the ACK ruri contains the same destination port which was used 
for the INVITE, otherwise a new connection will be created.

Best regards,

Federico

On Fri, May 18, 2018 at 2:31 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
Thank you Alex and Federico,

I verified, and SO_REUSEPORT is defined on my OS.  I am using Kamailio 5.2 and 
I set ‘tcp_reuse_port=yes;’ and $fs;  this has been to no avail as ‘ACK’s’ are 
still using the high port number randomly assigned by Kamailio.

Thank you all for sharing your knowledge!


From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Friday, May 18, 2018 12:57 AM
To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Transport issue thought

You are right Alex, Linux kernel didn't support SO_REUSEPORT, which allows a 
socket to be used as source for a tcp connection while is already bound, until 
version 3.9.
Kamailio's parameter tcp_reuse_port, if enabled and your OS has support for 
SO_REUSEPORT (so not only Linux but FreeBSD, OSX and others), allows you to use 
force_send_socket (or $fs) to send messages from a TCP port kamailio is 
listening to.

Cheers,

Federico

On Fri, May 18, 2018 at 12:28 AM, Alex Balashov 
<abalas...@evaristesys.com<mailto:abalas...@evaristesys.com>> wrote:
When an outgoing TCP connection is opened, either a port can be
explicitly bound, or it is auto-assigned by the OS's networking stack. I
believe Kamailio does the latter and does not offer options to constrain
the range. If it does, I'm not aware of any apart from this one:

https://www.kamailio.org/wiki/cookbooks/5.1.x/core#tcp_reuse_port

Not sure if it would help in this case, you'd have to give it a try.

On Thu, May 17, 2018 at 10:21:34PM +, Wilkins, Steve wrote:

> It appears that the bottom line of my TCP transport not working is that 
> Kamailio is randomly assigning large port numbers to send the TCP traffic out 
> on.
> I am not able to randomly open high ports for this purpose.  Is there a way 
> to tell Kamailio to only use specific ports for this.  I have tried using
> force_send_socket() with Kamailio' s IP, and the port I want to send out on, 
> but this did not work either.  Does this sound like I am on the right track?
>
> Thank you,
> -
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>
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--
Alex Balashov | Principal | Evariste Systems LLC

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

Re: [SR-Users] Transport issue thought

[Wilkins, Steve]

I have tried using t_relay_to_tcp() and t_realy, both with and without setting 
$ru = $ru + ";transport=tcp" right before the call.
I get different errors and failures depending on which way I make the call.  
Which log would you want to see?


  *   I also wanted to point out that if I use UDP, the call connects (it is a 
WebRTC client to WebRTC client call, and both clients are registered in 
Kamailio).  The call just gets dropped after 30 seconds because Asterisk 15.3 
is not receiving the ACK back to the ‘200 OK’.  Aas stated, TCP does not even 
connect.


  *   Another very interesting thing is that if I use UDP and Asterisk 14.6, 
the call works perfect! It stays connected and I have full duplex Audio and 
Video

Thank you!


From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Federico Cabiddu
Sent: Friday, May 18, 2018 8:49 AM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] Transport issue thought

Can you print the logs when it tries to send the ACK?
Check also that the ACK ruri contains the same destination port which was used 
for the INVITE, otherwise a new connection will be created.

Best regards,

Federico

On Fri, May 18, 2018 at 2:31 PM, Wilkins, Steve 
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
Thank you Alex and Federico,

I verified, and SO_REUSEPORT is defined on my OS.  I am using Kamailio 5.2 and 
I set ‘tcp_reuse_port=yes;’ and $fs;  this has been to no avail as ‘ACK’s’ are 
still using the high port number randomly assigned by Kamailio.

Thank you all for sharing your knowledge!


From: sr-users 
[mailto:sr-users-boun...@lists.kamailio.org<mailto:sr-users-boun...@lists.kamailio.org>]
 On Behalf Of Federico Cabiddu
Sent: Friday, May 18, 2018 12:57 AM
To: Kamailio (SER) - Users Mailing List 
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Transport issue thought

You are right Alex, Linux kernel didn't support SO_REUSEPORT, which allows a 
socket to be used as source for a tcp connection while is already bound, until 
version 3.9.
Kamailio's parameter tcp_reuse_port, if enabled and your OS has support for 
SO_REUSEPORT (so not only Linux but FreeBSD, OSX and others), allows you to use 
force_send_socket (or $fs) to send messages from a TCP port kamailio is 
listening to.

Cheers,

Federico

On Fri, May 18, 2018 at 12:28 AM, Alex Balashov 
<abalas...@evaristesys.com<mailto:abalas...@evaristesys.com>> wrote:
When an outgoing TCP connection is opened, either a port can be
explicitly bound, or it is auto-assigned by the OS's networking stack. I
believe Kamailio does the latter and does not offer options to constrain
the range. If it does, I'm not aware of any apart from this one:

https://www.kamailio.org/wiki/cookbooks/5.1.x/core#tcp_reuse_port

Not sure if it would help in this case, you'd have to give it a try.

On Thu, May 17, 2018 at 10:21:34PM +, Wilkins, Steve wrote:

> It appears that the bottom line of my TCP transport not working is that 
> Kamailio is randomly assigning large port numbers to send the TCP traffic out 
> on.
> I am not able to randomly open high ports for this purpose.  Is there a way 
> to tell Kamailio to only use specific ports for this.  I have tried using
> force_send_socket() with Kamailio' s IP, and the port I want to send out on, 
> but this did not work either.  Does this sound like I am on the right track?
>
> Thank you,
> -
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users


--
Alex Balashov | Principal | Evariste Systems LLC

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

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