subject:"\[OpenSIPS\-Users\] sdp"

Re: [OpenSIPS-Users] SDP

2024-03-22 Thread Bogdan-Andrei Iancu

On 2.2 you cannot use variables as parms for any script functions, some 
of them do not support variables. And replace_body() is one of them.


In 3.4 all script functions can take vars as params, so you should be fine.

Regards,

Bogdan-Andrei Iancu

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

On 22.03.2024 08:37, Stefan Carlsson wrote:
Still on 2.2 but are planning to migrste later in the spring to Ubuntu 
and 3.4   ..


Can you please in some quick notes tell me what must change in the 
script if required

Regards …

Stefan


*From:* Bogdan-Andrei Iancu 
*Sent:* Thursday, 21 March 2024 17:30
*To:* Stefan Carlsson; users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] SDP

Hi,

What OpenSIPS version are you using?

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com
On 21.03.2024 00:09, Stefan Carlsson wrote:


Hi !

I sincerely apologize that I not read thru all the modules before my 
orginal post , I’ve found that textops module can search and replace


the text in both the header and the body. (which I mailed to the 
requester)


I’ve tried some of the exported functions but:

orginal snap:  o=root 276479022 276479022 IN IP4 172.24.7.20

$avp(reinv_sess_id) = “12345678”;    # Got this from  
$avp(reinv_oline) = $(rb{sdp.line,o})   and some regex tweeking and 
 $(avp(reinv_result){s.select,1,;})


$avp(reinv_sess_ver) = “276479022”;

replace_body("$avp(reinv_sess_ver)","$avp(reinv_sess_id)");

This seemed not to work …

but this did:

Original text snap:    a=sendrecv

replace_body("a=sendrecv", "a=sendonly");   # Indicate call hold 
according RFC3264


// Regards …

Stefan

*From:* Bogdan-Andrei Iancu 
*Sent:* Wednesday, 20 March, 2024 14:44
*To:* OpenSIPS users mailling list ; Stefan 
Carlsson 

*Subject:* Re: [OpenSIPS-Users] SDP

Nothing is impossible as time as you search in the right place (aka 
the Manual). See my reply on your thread.


Regards,

Bogdan-Andrei Iancu
  
OpenSIPS Founder and Developer

   https://www.opensips-solutions.com
   https://www.siphub.com

On 14.03.2024 09:50, Stefan Carlsson wrote:

Hi !

You can use the module sipmsgops to view and change some of the
ADP fields such as codecs and streams … but unfortunately you
don’t have

complete control of the SDP compared to what it seems could be
done on the Kamailio SIP router

I need to change the a=sendrecv field and it seems to be impossible ☹

Regards …

_

Stefan Carlsson

*From:* Users 
<mailto:users-boun...@lists.opensips.org> *On Behalf Of *Prathibha B
*Sent:* Thursday, 14 March, 2024 05:16
*To:* OpenSIPS users mailling list 
<mailto:users@lists.opensips.org>
*Subject:* [OpenSIPS-Users] SDP

How to see the SDP in opensips?

-- 


Regards,

B.Prathibha



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Re: [OpenSIPS-Users] SDP

2024-03-21 Thread Prathibha B

3.4

Sent from Outlook for Android<https://aka.ms/AAb9ysg>

From: Users  on behalf of Bogdan-Andrei Iancu 

Sent: Thursday, March 21, 2024 9:59:51 PM
To: Stefan Carlsson ; users@lists.opensips.org 

Subject: Re: [OpenSIPS-Users] SDP

Hi,

What OpenSIPS version are you using?

Regards,

Bogdan-Andrei Iancu

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

On 21.03.2024 00:09, Stefan Carlsson wrote:

Hi !

I sincerely apologize that I not read thru all the modules before my orginal 
post , I’ve found that textops module can search and replace

the text in both the header and the body. (which I mailed to the requester)



I’ve tried some of the exported functions but:



orginal snap:  o=root 276479022 276479022 IN IP4 172.24.7.20





$avp(reinv_sess_id) = “12345678”;# Got this from  $avp(reinv_oline) = 
$(rb{sdp.line,o})   and some regex tweeking and  
$(avp(reinv_result){s.select,1,;})





$avp(reinv_sess_ver) = “276479022”;



replace_body("$avp(reinv_sess_ver)","$avp(reinv_sess_id)");



This seemed not to work …



but this did:



Original text snap:a=sendrecv



replace_body("a=sendrecv", "a=sendonly");   # Indicate call hold according 
RFC3264







// Regards …

Stefan





From: Bogdan-Andrei Iancu <mailto:bog...@opensips.org>
Sent: Wednesday, 20 March, 2024 14:44
To: OpenSIPS users mailling list 
<mailto:users@lists.opensips.org>; Stefan Carlsson 
<mailto:stefan.carls...@tetab.nu>
Subject: Re: [OpenSIPS-Users] SDP



Nothing is impossible as time as you search in the right place (aka the 
Manual). See my reply on your thread.

Regards,


Bogdan-Andrei Iancu



OpenSIPS Founder and Developer

  https://www.opensips-solutions.com

  https://www.siphub.com

On 14.03.2024 09:50, Stefan Carlsson wrote:

Hi !



You can use the module sipmsgops to view and change some of the ADP fields such 
as codecs and streams … but unfortunately you don’t have

complete control of the SDP compared to what it seems could be done on the 
Kamailio SIP router



I need to change the a=sendrecv field and it seems to be impossible  ☹



Regards …

_

Stefan Carlsson



From: Users 
<mailto:users-boun...@lists.opensips.org> On 
Behalf Of Prathibha B
Sent: Thursday, 14 March, 2024 05:16
To: OpenSIPS users mailling list 
<mailto:users@lists.opensips.org>
Subject: [OpenSIPS-Users] SDP



How to see the SDP in opensips?



--

Regards,

B.Prathibha



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Re: [OpenSIPS-Users] SDP

2024-03-21 Thread Bogdan-Andrei Iancu


Hi,

What OpenSIPS version are you using?

Regards,

Bogdan-Andrei Iancu

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

On 21.03.2024 00:09, Stefan Carlsson wrote:


Hi !

I sincerely apologize that I not read thru all the modules before my 
orginal post , I’ve found that textops module can search and replace


the text in both the header and the body. (which I mailed to the 
requester)


I’ve tried some of the exported functions but:

orginal snap:  o=root 276479022 276479022 IN IP4 172.24.7.20

$avp(reinv_sess_id) = “12345678”;    # Got this from  
$avp(reinv_oline) = $(rb{sdp.line,o})   and some regex tweeking and 
 $(avp(reinv_result){s.select,1,;})


$avp(reinv_sess_ver) = “276479022”;

replace_body("$avp(reinv_sess_ver)","$avp(reinv_sess_id)");

This seemed not to work …

but this did:

Original text snap:    a=sendrecv

replace_body("a=sendrecv", "a=sendonly");   # Indicate call hold 
according RFC3264


// Regards …

Stefan

*From:* Bogdan-Andrei Iancu 
*Sent:* Wednesday, 20 March, 2024 14:44
*To:* OpenSIPS users mailling list ; Stefan 
Carlsson 

*Subject:* Re: [OpenSIPS-Users] SDP

Nothing is impossible as time as you search in the right place (aka 
the Manual). See my reply on your thread.


Regards,

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

On 14.03.2024 09:50, Stefan Carlsson wrote:

Hi !

You can use the module sipmsgops to view and change some of the
ADP fields such as codecs and streams … but unfortunately you
don’t have

complete control of the SDP compared to what it seems could be
done on the Kamailio SIP router

I need to change the a=sendrecv field and it seems to be impossible ☹

Regards …

_

Stefan Carlsson

*From:* Users 
<mailto:users-boun...@lists.opensips.org> *On Behalf Of *Prathibha B
*Sent:* Thursday, 14 March, 2024 05:16
*To:* OpenSIPS users mailling list 
<mailto:users@lists.opensips.org>
*Subject:* [OpenSIPS-Users] SDP

How to see the SDP in opensips?

-- 


Regards,

B.Prathibha



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Re: [OpenSIPS-Users] SDP

2024-03-20 Thread Bogdan-Andrei Iancu

Nothing is impossible as time as you search in the right place (aka the 
Manual). See my reply on your thread.


Regards,

Bogdan-Andrei Iancu

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

On 14.03.2024 09:50, Stefan Carlsson wrote:


Hi !

You can use the module sipmsgops to view and change some of the ADP 
fields such as codecs and streams … but unfortunately you don’t have


complete control of the SDP compared to what it seems could be done on 
the Kamailio SIP router


I need to change the a=sendrecv field and it seems to be impossible ☹

Regards …

_

Stefan Carlsson

*From:* Users  *On Behalf Of 
*Prathibha B

*Sent:* Thursday, 14 March, 2024 05:16
*To:* OpenSIPS users mailling list 
*Subject:* [OpenSIPS-Users] SDP

How to see the SDP in opensips?

--

Regards,

B.Prathibha


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Re: [OpenSIPS-Users] SDP

2024-03-20 Thread Bogdan-Andrei Iancu


The online Manual should be your Bible here ;)

https://opensips.org/Documentation/Script-CoreVar-3-4#rb

Regards,

Bogdan-Andrei Iancu

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

On 14.03.2024 06:16, Prathibha B wrote:

How to see the SDP in opensips?

--
Regards,
B.Prathibha

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Re: [OpenSIPS-Users] SDP

2024-03-14 Thread Stefan Carlsson

Hi !

You can use the module sipmsgops to view and change some of the ADP fields such 
as codecs and streams … but unfortunately you don’t have
complete control of the SDP compared to what it seems could be done on the 
Kamailio SIP router

I need to change the a=sendrecv field and it seems to be impossible  ☹

Regards …
_
Stefan Carlsson

From: Users  On Behalf Of Prathibha B
Sent: Thursday, 14 March, 2024 05:16
To: OpenSIPS users mailling list 
Subject: [OpenSIPS-Users] SDP

How to see the SDP in opensips?

--
Regards,
B.Prathibha
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[OpenSIPS-Users] SDP

2024-03-13 Thread Prathibha B

How to see the SDP in opensips?

-- 
Regards,
B.Prathibha
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Re: [OpenSIPS-Users] SDP reroute and rewrite

2024-02-12 Thread M S

Hi Igor,
You are absolutely right. For anybody reading this in future, I figured it
works from branch_route because as Igor explained, that's where you have
access to unaltered packet. Thanks!

On Mon, Feb 12, 2024 at 10:35 AM Ihor Olkhovskyi 
wrote:

> Usually you work in failure_route with packet that was in a state when you
> called t_relay.
>
> Best practice would be to work with rtp-related procedures in branch
> routes, so you will get in failure_rroute unaltered packet, before
> rtp_offer manipulations from previous time.
>
> Cheers,
> Ihor
>
> Le sam. 10 févr. 2024 à 03:03, M S  a écrit :
>
>> Hi list,
>> When using rtpproxy_offer/answer, how can I rewrite SDP media IP/port if
>> for example the first route rejects the calls and I have to send the call
>> (in failure_route) to the next destination (where second destination uses
>> different media ip/port/rtpproxy set)?
>> If I just call rtpproxy_offer again, on second try it changes IPs to
>> something like:
>> m=audio 1106210576 RTP/AVP 8 0 101
>> c=IN IP4 172.17.182.213172.17.182.210
>> a=rtcp:11063 IN IP4 172.17.182.21310577 IN IP4 172.17.182.210
>>
>> I tried unforce and it didn't work.
>> Thanks!
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Best regards,
> Ihor (Igor)
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Re: [OpenSIPS-Users] SDP reroute and rewrite

2024-02-12 Thread Ihor Olkhovskyi

Usually you work in failure_route with packet that was in a state when you
called t_relay.

Best practice would be to work with rtp-related procedures in branch
routes, so you will get in failure_rroute unaltered packet, before
rtp_offer manipulations from previous time.

Cheers,
Ihor

Le sam. 10 févr. 2024 à 03:03, M S  a écrit :

> Hi list,
> When using rtpproxy_offer/answer, how can I rewrite SDP media IP/port if
> for example the first route rejects the calls and I have to send the call
> (in failure_route) to the next destination (where second destination uses
> different media ip/port/rtpproxy set)?
> If I just call rtpproxy_offer again, on second try it changes IPs to
> something like:
> m=audio 1106210576 RTP/AVP 8 0 101
> c=IN IP4 172.17.182.213172.17.182.210
> a=rtcp:11063 IN IP4 172.17.182.21310577 IN IP4 172.17.182.210
>
> I tried unforce and it didn't work.
> Thanks!
> ___
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> Users@lists.opensips.org
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>


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Ihor (Igor)
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[OpenSIPS-Users] SDP reroute and rewrite

2024-02-09 Thread M S

Hi list,
When using rtpproxy_offer/answer, how can I rewrite SDP media IP/port if
for example the first route rejects the calls and I have to send the call
(in failure_route) to the next destination (where second destination uses
different media ip/port/rtpproxy set)?
If I just call rtpproxy_offer again, on second try it changes IPs to
something like:
m=audio 1106210576 RTP/AVP 8 0 101
c=IN IP4 172.17.182.213172.17.182.210
a=rtcp:11063 IN IP4 172.17.182.21310577 IN IP4 172.17.182.210

I tried unforce and it didn't work.
Thanks!
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Re: [OpenSIPS-Users] SDP manipulation & rtpengine

2019-03-07 Thread Bogdan-Andrei Iancu

Hmmm, indeed, passing SDP as variable to rtpengine is possible only with 
3.0 , my badtoo many versions.


This is the commit you are looking for:
https://github.com/OpenSIPS/opensips/commit/a27797d04ba418eb3ac4c2d6dadd0cdf7f3c17b6#diff-069ca6b59f3521936eca7bf3684ae63d

maybe you can grab the module from 3.0 right after that commit and copy 
it in 2.4 tree - it may compile.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 03/06/2019 10:47 PM, Pat Burke wrote:


Thanks Bogdan,


I see that it is part of 3.0, but not a part of 2.4.  Is this a module 
we can pull back to 2.4?



Regards,
*Pat Burke*

Voxtelesys | solutions to grow your business
__
Direct: (402) 403-5121 |   Cell: (402) 443-8929 |   Email: 
p...@voxtelesys.com <mailto:p...@voxtelesys.com>

1801 23rd Avenue North |  Suite 217 |  Fargo, North Dakota 58102


-Original Message-
From: "Bogdan-Andrei Iancu" mailto:bog...@opensips.org>>
To: "OpenSIPS users mailling list" mailto:users@lists.opensips.org>>, "Pat Burke"
mailto:p...@voxtelesys.com>>
    Date: 03/05/19 09:45
Subject: Re: [OpenSIPS-Users] SDP manipulation & rtpengine

Hi Pat,

What you can do is to grab the SDP from the msg into a variable,
to do whatever fixes/change you have to directly in the variable
and push the body via variable to rtpengine.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.comOpenSIPS Summit 2019
   https://www.opensips.org/events/Summit-2019Amsterdam/

On 02/27/2019 11:30 PM, Pat Burke wrote:


Hello:


I am using trying to manipulate parts of the SDP body before
calling rtpengine_offer / rtpengine_answer. However, any changes
made via textops functions such as subst_body, replace_body,
replace_body_all, etc. do not seem to impact the SDP that is sent
to rtpengine.

In my particular case, rtpengine fails to parse the SDP because
of an extra carriage return line feed sent in the SDP. Is there a
way to send rtpengine manipulated SDP, rather than just the SDP
sent in the request?

*_Use case:_*
if (subst_body("/(^a=.*\r\n)\r\n/\1/g")) {
xlog("L_INFO", "bad SDP --- duplicate CRLF");
}

rtpengine_offer(" ... options ... ");


Regards,
*Pat Burke*

Voxtelesys | solutions to grow your business

__
Direct: (402) 403-5121 | Cell: (402) 443-8929 | Email:
p...@voxtelesys.com <mailto:p...@voxtelesys.com>
1801 23rd Avenue North | Suite 217 | Fargo, North Dakota 58102


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Re: [OpenSIPS-Users] SDP manipulation & rtpengine

2019-03-06 Thread Pat Burke


Thanks Bogdan,




I see that it is part of 3.0, but not a part of 2.4.  Is this a module we can 
pull back to 2.4?



Regards,
Pat Burke



__
Direct: (402) 403-5121   |   Cell: (402) 443-8929  |   Email: 
p...@voxtelesys.com
1801 23rd Avenue North   |  Suite 217    |  Fargo, North Dakota 58102
 

-Original Message-
From: "Bogdan-Andrei Iancu" 
To: "OpenSIPS users mailling list" , "Pat Burke" 

Date: 03/05/19 09:45
Subject: Re: [OpenSIPS-Users] SDP manipulation & rtpengine

Hi Pat,

What you can do is to grab the SDP from the msg into a variable, to do whatever 
fixes/change you have to directly in the variable and push the body via 
variable to rtpengine.

Regards,

Bogdan-Andrei Iancu OpenSIPS Founder and Developer   
https://www.opensips-solutions.comOpenSIPS Summit 2019   
https://www.opensips.org/events/Summit-2019Amsterdam/
On 02/27/2019 11:30 PM, Pat Burke wrote:


Hello:


I am using trying to manipulate parts of the SDP body before calling 
rtpengine_offer / rtpengine_answer.  However, any changes made via textops 
functions such as subst_body, replace_body, replace_body_all, etc. do not seem 
to impact the SDP that is sent to rtpengine.


In my particular case, rtpengine fails to parse the SDP because of an extra 
carriage return line feed sent in the SDP.  Is there a way to send rtpengine 
manipulated SDP, rather than just the SDP sent in the request?


Use case:
if (subst_body("/(^a=.*\r\n)\r\n/\1/g")) {
  xlog("L_INFO", "bad SDP --- duplicate CRLF");
}


rtpengine_offer(" ... options ... ");



Regards,
Pat Burke



__
Direct: (402) 403-5121   |   Cell: (402) 443-8929  |   Email: 
p...@voxtelesys.com
1801 23rd Avenue North   |  Suite 217|  Fargo, North Dakota 58102
 



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Re: [OpenSIPS-Users] SDP manipulation & rtpengine

2019-03-05 Thread Bogdan-Andrei Iancu


Hi Pat,

What you can do is to grab the SDP from the msg into a variable, to do 
whatever fixes/change you have to directly in the variable and push the 
body via variable to rtpengine.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 02/27/2019 11:30 PM, Pat Burke wrote:


Hello:


I am using trying to manipulate parts of the SDP body before calling 
rtpengine_offer / rtpengine_answer.  However, any changes made via 
textops functions such as subst_body, replace_body, replace_body_all, 
etc. do not seem to impact the SDP that is sent to rtpengine.


In my particular case, rtpengine fails to parse the SDP because of an 
extra carriage return line feed sent in the SDP.  Is there a way to 
send rtpengine manipulated SDP, rather than just the SDP sent in the 
request?


*_Use case:_*
if (subst_body("/(^a=.*\r\n)\r\n/\1/g")) {
xlog("L_INFO", "bad SDP --- duplicate CRLF");
}

rtpengine_offer(" ... options ... ");


Regards,
*Pat Burke*

Voxtelesys | solutions to grow your business
__
Direct: (402) 403-5121 |   Cell: (402) 443-8929 |   Email: 
p...@voxtelesys.com 

1801 23rd Avenue North |  Suite 217 |  Fargo, North Dakota 58102


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[OpenSIPS-Users] SDP manipulation & rtpengine

2019-02-27 Thread Pat Burke


Hello:


I am using trying to manipulate parts of the SDP body before calling 
rtpengine_offer / rtpengine_answer.  However, any changes made via textops 
functions such as subst_body, replace_body, replace_body_all, etc. do not seem 
to impact the SDP that is sent to rtpengine.


In my particular case, rtpengine fails to parse the SDP because of an extra 
carriage return line feed sent in the SDP.  Is there a way to send rtpengine 
manipulated SDP, rather than just the SDP sent in the request?


Use case:
if (subst_body("/(^a=.*\r\n)\r\n/\1/g")) {
  xlog("L_INFO", "bad SDP --- duplicate CRLF");
}


rtpengine_offer(" ... options ... ");


Regards,
Pat Burke



__
Direct: (402) 403-5121   |   Cell: (402) 443-8929  |   Email: 
p...@voxtelesys.com
1801 23rd Avenue North   |  Suite 217    |  Fargo, North Dakota 58102
 

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Re: [OpenSIPS-Users] SDP and ptime, maxptime removal

2018-10-30 Thread Stefan Carlsson

Ok !

Thanks ...

But I have three of them after :"codec_delete_except_re("PCMA") ;   ",  
see below snap.



t=0 0
m=audio 46056 RTP/AVP 8
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=ptime:20
a=maxptime:40
a=ptime:30
a=maxptime:30



Btw, thanks for your help and reply.


Kind Regards / Vänligen …

Stefan Carlsson

-Original Message-
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan 
Crainea
Sent: Monday, October 29, 2018 16:35
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] SDP and ptime, maxptime removal

Hi, Stefan!

There is no bug here - ptime and maxptime are not associated to a codec, but to 
the entire media session. So it should not be removed if you delete a certain.

Best regards,
Razvan

On 10/26/18 3:00 PM, Stefan Carlsson wrote:
> Hi !
> We have a serious issue that force us to remove all codecs except 
> PCMA.   We use: codec_delete_except_re("PCMA");  from the 
> sipmsgops module but the function doesn’t removes the associated 
> ptime, maxptime.   How can we also remove this, any ideas or is it a 
> bug in the sipmsgops  module?
> We use version  2.2.6   on a Centos machine.
> Kind Regards
> Stefan
> 
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Re: [OpenSIPS-Users] SDP and ptime, maxptime removal

2018-10-29 Thread Răzvan Crainea


Hi, Stefan!

There is no bug here - ptime and maxptime are not associated to a codec, 
but to the entire media session. So it should not be removed if you 
delete a certain.


Best regards,
Razvan

On 10/26/18 3:00 PM, Stefan Carlsson wrote:

Hi !
We have a serious issue that force us to remove all codecs except 
PCMA.   We use: codec_delete_except_re("PCMA");  from the sipmsgops 
module
but the function doesn’t removes the associated ptime, maxptime.   How 
can we also remove this, any ideas or is it a bug in the sipmsgops  module?

We use version  2.2.6   on a Centos machine.
Kind Regards
Stefan

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  http://www.opensips-solutions.com

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[OpenSIPS-Users] SDP and ptime, maxptime removal

2018-10-26 Thread Stefan Carlsson

Hi !

We have a serious issue that force us to remove all codecs except PCMA.   We 
use: codec_delete_except_re("PCMA");  from the sipmsgops module
but the function doesn't removes the associated ptime, maxptime.   How can we 
also remove this, any ideas or is it a bug in the sipmsgops  module?

We use version  2.2.6   on a Centos machine.


Kind Regards

Stefan



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Re: [OpenSIPS-Users] SDP version increment without a change in SDP

2018-01-24 Thread Schoolhouse Filing


Ben,

Thank you that is so helpful.

Adrian.


On 24/01/18 19:35, Ben Newlin wrote:


Adrian,

I can’t help with a way not to increment the version number on 
RTPProxy, but I believe I know why the provider is not happy with the 
exchange.


The problem is not that the version number has incremented without a 
change to the SDP. As you have pointed out, RFC 4566 does not prohibit 
changing the version number and I think the provider was wrong to 
point you to that RFC. I think it’s likely the issue isn’t with the 
SDP itself, but with the exchange. It is covered in RFC 3261, Section 
13.2.1, specifically these two bullets:


·If the initial offer is in an INVITE, the answer MUST be in a

 reliable non-failure message from UAS back to UAC which is

 correlated to that INVITE. For this specification, that is

 only the final 2xx response to that INVITE.  That same exact

 answer MAY also be placed in any provisional responses sent

 prior to the answer.  The UAC MUST treat the first session

 description it receives as the answer, and MUST ignore any

 session descriptions in subsequent responses to the initial

 INVITE.

·Once the UAS has sent or received an answer to the initial
 offer, it MUST NOT generate subsequent offers in any responses
 to the initial INVITE.  This means that a UAS based on this
 specification alone can never generate subsequent offers until
 completion of the initial transaction.

So the problem is that the SDP is changing between the 183 and 200 OK, 
even though it is only the version. When providing SDP in an 
unreliable response, the SDP in the final response is required to be 
exactly the same. You are not allowed to change the SDP once it has 
been sent without initiating a new offer/answer, which you can’t do in 
the 200 OK as you haven’t completed the previous exchange with a final 
response.


Having said that, this is not a widely enforced rule. Nearly all SIP 
implementations are tolerant of this or at least will simply ignore 
SDP’s after the first without terminating the call. Many even support 
receiving multiple different SDP’s in multiple 183 responses during a 
single INVITE exchange. But I have worked with some who do not 
tolerate it and unfortunately the RFC is on their side.


Of course, only your provider can confirm if this is, in fact, their 
issue with the exchange.


Thanks,

Ben

*From: *Users  on behalf of Adrian 
Fretwell 

*Reply-To: *OpenSIPS users mailling list 
*Date: *Wednesday, January 24, 2018 at 2:06 PM
*To: *OpenSIPS users mailling list 
*Subject: *[OpenSIPS-Users] SDP version increment without a change in SDP

Hello All,  I have an issue for which I have not been able to find a 
definitive answer in the RFCs.


I use RTPProxy as part of NAT traversal so RTP streams are set up 
between upstream provider and my proxy, and between my proxy and the UAC.


The problem I have, is UAC changes it's RTP port between 183 and 200 
OK and thus increments the SDP version number, BUT the ports DO NOT 
change between proxy and upstream provider but the SDP version number 
does, because it is passed through.  The upstream provider says this 
violates RFC 4566 and sends an immediate BYE after the final ACK.


I read that RFC 4566 says:

" / is a version number for this session description.  Its
  usage is up to the creating tool, so long as  is
  increased when a modification is made to the session data.  Again,
  it is RECOMMENDED that an NTP format timestamp is used/."

The problem is I can't find an RFC stating "You MUST NOT increment
version number if no change is made to the SDP".

So I must either:

1. Prove that it should be OK to increment the version number without 
any change to the SDP.  Or


2. Find a way of not passing through the incremented version number to 
the upstream side where the SDP has not actually changed.


I hope that makes sense...  As always any help very nuch appreciated.

Kind regards,

Adrian Fretwell
Nottinghamshire
UK



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Re: [OpenSIPS-Users] SDP version increment without a change in SDP

2018-01-24 Thread Ben Newlin

Adrian,

I can’t help with a way not to increment the version number on RTPProxy, but I 
believe I know why the provider is not happy with the exchange.

The problem is not that the version number has incremented without a change to 
the SDP. As you have pointed out, RFC 4566 does not prohibit changing the 
version number and I think the provider was wrong to point you to that RFC. I 
think it’s likely the issue isn’t with the SDP itself, but with the exchange. 
It is covered in RFC 3261, Section 13.2.1, specifically these two bullets:


· If the initial offer is in an INVITE, the answer MUST be in a
 reliable non-failure message from UAS back to UAC which is
 correlated to that INVITE.  For this specification, that is
 only the final 2xx response to that INVITE.  That same exact
 answer MAY also be placed in any provisional responses sent
 prior to the answer.  The UAC MUST treat the first session
 description it receives as the answer, and MUST ignore any
 session descriptions in subsequent responses to the initial
 INVITE.

· Once the UAS has sent or received an answer to the initial

 offer, it MUST NOT generate subsequent offers in any responses

 to the initial INVITE.  This means that a UAS based on this

 specification alone can never generate subsequent offers until

 completion of the initial transaction.

So the problem is that the SDP is changing between the 183 and 200 OK, even 
though it is only the version. When providing SDP in an unreliable response, 
the SDP in the final response is required to be exactly the same. You are not 
allowed to change the SDP once it has been sent without initiating a new 
offer/answer, which you can’t do in the 200 OK as you haven’t completed the 
previous exchange with a final response.

Having said that, this is not a widely enforced rule. Nearly all SIP 
implementations are tolerant of this or at least will simply ignore SDP’s after 
the first without terminating the call. Many even support receiving multiple 
different SDP’s in multiple 183 responses during a single INVITE exchange. But 
I have worked with some who do not tolerate it and unfortunately the RFC is on 
their side.

Of course, only your provider can confirm if this is, in fact, their issue with 
the exchange.

Thanks,
Ben



From: Users  on behalf of Adrian Fretwell 

Reply-To: OpenSIPS users mailling list 
Date: Wednesday, January 24, 2018 at 2:06 PM
To: OpenSIPS users mailling list 
Subject: [OpenSIPS-Users] SDP version increment without a change in SDP


Hello All,  I have an issue for which I have not been able to find a definitive 
answer in the RFCs.

I use RTPProxy as part of NAT traversal so RTP streams are set up between 
upstream provider and my proxy, and between my proxy and the UAC.

The problem I have, is UAC changes it's RTP port between 183 and 200 OK and 
thus increments the SDP version number, BUT the ports DO NOT change between 
proxy and upstream provider but the SDP version number does, because it is 
passed through.  The upstream provider says this violates RFC 4566 and sends an 
immediate BYE after the final ACK.

I read that RFC 4566 says:

"  is a version number for this session description.  Its
  usage is up to the creating tool, so long as  is
  increased when a modification is made to the session data.  Again,
  it is RECOMMENDED that an NTP format timestamp is used."

The problem is I can't find an RFC stating "You MUST NOT increment
version number if no change is made to the SDP".

So I must either:

1. Prove that it should be OK to increment the version number without any 
change to the SDP.  Or

2. Find a way of not passing through the incremented version number to the 
upstream side where the SDP has not actually changed.

I hope that makes sense...  As always any help very nuch appreciated.
Kind regards,

Adrian Fretwell
Nottinghamshire
UK
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[OpenSIPS-Users] SDP version increment without a change in SDP

2018-01-24 Thread Adrian Fretwell

Hello All,  I have an issue for which I have not been able to find a 
definitive answer in the RFCs.


I use RTPProxy as part of NAT traversal so RTP streams are set up 
between upstream provider and my proxy, and between my proxy and the UAC.


The problem I have, is UAC changes it's RTP port between 183 and 200 OK 
and thus increments the SDP version number, BUT the ports DO NOT change 
between proxy and upstream provider but the SDP version number does, 
because it is passed through.  The upstream provider says this violates 
RFC 4566 and sends an immediate BYE after the final ACK.


I read that RFC 4566 says:

" / is a version number for this session description.  Its//
//  usage is up to the creating tool, so long as  is//
//  increased when a modification is made to the session data.  Again,//
//  it is RECOMMENDED that an NTP format timestamp is used/."

The problem is I can't find an RFC stating "You MUST NOT increment
version number if no change is made to the SDP".

So I must either:

1. Prove that it should be OK to increment the version number without 
any change to the SDP.  Or


2. Find a way of not passing through the incremented version number to 
the upstream side where the SDP has not actually changed.


I hope that makes sense...  As always any help very nuch appreciated.

Kind regards,

Adrian Fretwell
Nottinghamshire
UK
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Re: [OpenSIPS-Users] sdp

2017-05-22 Thread Bogdan-Andrei Iancu


Hi,

If you have an re-INVITE with active media, you just have to re-insert 
the rtpengine, exactly as you did it for the initial INVITE. This will 
properly handle the on-hold resume.


Regards,

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

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 05/21/2017 04:24 AM, volga...@networklab.ca wrote:

Hello Bogdan-Andrei,
The issue with on hold RTP stream resume. I am not sure how to 
opensips should handle properly.

I code bellow provide partial solution.

volga629

On Tue, 16 May, 2017 at 4:38 AM, Bogdan-Andrei Iancu 
 wrote:

Hello Volga,

What exactly does not work for you ? the detection at SIP level of 
the hold resume ? the actual RTP resume ?


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

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html
On 05/15/2017 06:06 PM, volga...@networklab.ca wrote:

Hello Everyone,
Thank you

That extract ip correctly.

$var(cline-ip) = $(rb{sdp.line,c}{s.select,2, });

My issue that I am trying make opensips handle music on hold resume 
working properly.

I am not sure if possible do simpler way.


Relevant code

route[ONHOLD] {
if(is_method("INVITE|UPDATE") && has_body("application/sdp")) {
if(is_audio_on_hold()) {
if(search_body("a=sendonly") || 
search_body("a=inactive.")) {
$var(cline-ip) = 
$(rb{sdp.line,c}{s.select,2, });
xlog("L_INFO", "[$rm] On hold call 
SDP IP [$var(cline-ip)]\n");
if(!nat_uac_test("8") || 
$(var(cline-ip){s.select,1,.})==0) {
xlog("L_INFO", "[$rm] On 
hold call going from WAN SouceIP <$si> to LAN.\n");

rtpengine_offer("replace-origin external internal RTP/AVP ICE=remove");
t_on_reply("2");
}
}
}
}
}



volga629

On Mon, 15 May, 2017 at 5:23 AM, Bogdan-Andrei Iancu 
 wrote:

Hi Volga,

You can use the sdp transformation :
http://www.opensips.org/Documentation/Script-Tran-2-3#toc80

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

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html
On 05/14/2017 06:28 AM, volga...@networklab.ca wrote:

Hello Everyone,
What good approach to test/detect 0.0.0.0 in sdp c IN = line ?
I don't see any functions to parse sdp properly.

volga629


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Re: [OpenSIPS-Users] sdp

2017-05-20 Thread volga629


Hello Bogdan-Andrei,
The issue with on hold RTP stream resume. I am not sure how to opensips 
should handle properly.

I code bellow provide partial solution.

volga629

On Tue, 16 May, 2017 at 4:38 AM, Bogdan-Andrei Iancu 
 wrote:

Hello Volga,

What exactly does not work for you ? the detection at SIP level of 
the hold resume ? the actual RTP resume ?


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

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html
On 05/15/2017 06:06 PM, volga...@networklab.ca wrote:

Hello Everyone,
Thank you

That extract ip correctly.

$var(cline-ip) = $(rb{sdp.line,c}{s.select,2, });

My issue that I am trying make opensips handle music on hold resume 
working properly.

I am not sure if possible do simpler way.


Relevant code

route[ONHOLD] {
if(is_method("INVITE|UPDATE") && 
has_body("application/sdp")) {

if(is_audio_on_hold()) {
if(search_body("a=sendonly") || 
search_body("a=inactive.")) {
$var(cline-ip) = 
$(rb{sdp.line,c}{s.select,2, });
xlog("L_INFO", "[$rm] On hold call 
SDP IP [$var(cline-ip)]\n");
if(!nat_uac_test("8") || 
$(var(cline-ip){s.select,1,.})==0) {
xlog("L_INFO", "[$rm] On 
hold call going from WAN SouceIP <$si> to LAN.\n");

rtpengine_offer("replace-origin external internal RTP/AVP 
ICE=remove");

t_on_reply("2");
}
}
}
}
}



volga629

On Mon, 15 May, 2017 at 5:23 AM, Bogdan-Andrei Iancu 
 wrote:

Hi Volga,

You can use the sdp transformation :
http://www.opensips.org/Documentation/Script-Tran-2-3#toc80

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

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html
On 05/14/2017 06:28 AM, volga...@networklab.ca wrote:

Hello Everyone,
What good approach to test/detect 0.0.0.0 in sdp c IN = line ?
I don't see any functions to parse sdp properly.

volga629


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Re: [OpenSIPS-Users] sdp

2017-05-16 Thread Bogdan-Andrei Iancu


Hello Volga,

What exactly does not work for you ? the detection at SIP level of the 
hold resume ? the actual RTP resume ?


Best regards,

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

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 05/15/2017 06:06 PM, volga...@networklab.ca wrote:

Hello Everyone,
Thank you

That extract ip correctly.

$var(cline-ip) = $(rb{sdp.line,c}{s.select,2, });

My issue that I am trying make opensips handle music on hold resume 
working properly.

I am not sure if possible do simpler way.


Relevant code

route[ONHOLD] {
if(is_method("INVITE|UPDATE") && has_body("application/sdp")) {
if(is_audio_on_hold()) {
if(search_body("a=sendonly") || 
search_body("a=inactive.")) {
$var(cline-ip) = 
$(rb{sdp.line,c}{s.select,2, });
xlog("L_INFO", "[$rm] On hold call SDP 
IP [$var(cline-ip)]\n");
if(!nat_uac_test("8") || 
$(var(cline-ip){s.select,1,.})==0) {
xlog("L_INFO", "[$rm] On hold 
call going from WAN SouceIP <$si> to LAN.\n");

rtpengine_offer("replace-origin external internal RTP/AVP ICE=remove");
t_on_reply("2");
}
}
}
}
}



volga629

On Mon, 15 May, 2017 at 5:23 AM, Bogdan-Andrei Iancu 
 wrote:

Hi Volga,

You can use the sdp transformation :
http://www.opensips.org/Documentation/Script-Tran-2-3#toc80

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

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html
On 05/14/2017 06:28 AM, volga...@networklab.ca wrote:

Hello Everyone,
What good approach to test/detect 0.0.0.0 in sdp c IN = line ?
I don't see any functions to parse sdp properly.

volga629


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Re: [OpenSIPS-Users] sdp

2017-05-15 Thread volga629


Hello Everyone,
Thank you

That extract ip correctly.

$var(cline-ip) = $(rb{sdp.line,c}{s.select,2, });

My issue that I am trying make opensips handle music on hold resume 
working properly.

I am not sure if possible do simpler way.


Relevant code

route[ONHOLD] {
   if(is_method("INVITE|UPDATE") && has_body("application/sdp")) {
   if(is_audio_on_hold()) {
   if(search_body("a=sendonly") || 
search_body("a=inactive.")) {
   $var(cline-ip) = 
$(rb{sdp.line,c}{s.select,2, });
   xlog("L_INFO", "[$rm] On hold call SDP 
IP [$var(cline-ip)]\n");
   if(!nat_uac_test("8") || 
$(var(cline-ip){s.select,1,.})==0) {
   xlog("L_INFO", "[$rm] On hold 
call going from WAN SouceIP <$si> to LAN.\n");
   rtpengine_offer("replace-origin 
external internal RTP/AVP ICE=remove");

   t_on_reply("2");
   }
   }
   }
   }
}



volga629

On Mon, 15 May, 2017 at 5:23 AM, Bogdan-Andrei Iancu 
 wrote:

Hi Volga,

You can use the sdp transformation :
http://www.opensips.org/Documentation/Script-Tran-2-3#toc80

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

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html
On 05/14/2017 06:28 AM, volga...@networklab.ca wrote:

Hello Everyone,
What good approach to test/detect 0.0.0.0 in sdp c IN = line ?
I don't see any functions to parse sdp properly.

volga629


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Re: [OpenSIPS-Users] sdp

2017-05-15 Thread Bogdan-Andrei Iancu


Hi Volga,

You can use the sdp transformation :
http://www.opensips.org/Documentation/Script-Tran-2-3#toc80

Regards,

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

OpenSIPS Summit May 2017 Amsterdam
  http://www.opensips.org/events/Summit-2017Amsterdam.html

On 05/14/2017 06:28 AM, volga...@networklab.ca wrote:

Hello Everyone,
What good approach to test/detect 0.0.0.0 in sdp c IN = line ?
I don't see any functions to parse sdp properly.

volga629


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Re: [OpenSIPS-Users] sdp

2017-05-15 Thread Răzvan Crainea


Hi, Volga!

Check the SDP transformations[1].

[1] http://www.opensips.org/Documentation/Script-Tran-2-3#toc80

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 05/14/2017 06:28 AM, volga...@networklab.ca wrote:

Hello Everyone,
What good approach to test/detect 0.0.0.0 in sdp c IN = line ?
I don't see any functions to parse sdp properly.

volga629


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[OpenSIPS-Users] sdp

2017-05-13 Thread volga629


Hello Everyone,
What good approach to test/detect 0.0.0.0 in sdp c IN = line ?
I don't see any functions to parse sdp properly.

volga629
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Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.6

2016-03-31 Thread Louis Rochon

Anybody have any ideas?


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
Sent: Thursday, March 24, 2016 3:16 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

I recompiled with the 1.11.6, and did not resolve the issue. Traces show 
behaviour identical between 1.11.5 and 1.11.6.

Looking at the release notes for 1.11.6, there are some fixes that are closely 
related, but don't fix my specific issue. i.e. 95f5f79, 26a0a62. 

Any suggestions? Is there some yet-to-be documented B2BLogic module parameter 
that needs to be set? 

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
Sent: Monday, March 21, 2016 2:22 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

Thanks Liviu,

I will recompile and give it a try.

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Monday, March 21, 2016 9:31 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

Hi Louis,

There was a similar discussion back in August [1], concluding with a fix [2], 
made between the 1.11.5 and 1.11.6 releases. Updating to the latest OpenSIPS 
1.11 will most likely solve this problem.

[1]: http://lists.opensips.org/pipermail/users/2015-August/032239.html
[2]: https://github.com/OpenSIPS/opensips/commit/95f5f79b9250

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 21.03.2016 14:55, Louis Rochon wrote:
> Anybody?
>
> Louis
>
>
> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
> Sent: Wednesday, March 16, 2016 10:02 AM
> To: users@lists.opensips.org
> Subject: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in
> 1.11.5
>
> SDP in ACK lost in OpenSIPS 1.11.5
>
> This is something that used to work in OpenSIPS 1.8.1, but seems to have been 
> broken 1.11.5.
>
> Using the B2BUA facilities, I make the second leg of the call using 
> b2b_init_request. Then, if required, I move the call to another user agent 
> via a bash shell script, which issues a opensipsctl fifo b2b_bridge.
>
> For this in-dialog reinvite to work, there has to be an SDP in the ACK, which 
> is missing the trace produced using OpenSIPS 1.11.5. It's there in 1.8.1.
>
> Anyway of reintroducing the SDP in the ACK in 1.11.5?
>
> Trace Explanation
> -
> Objects: 10.10.8.103: source of initial call. Recipipient of reinvite.
>   10.10.10.205: CentOS running OpenSIPS with B2BUA. In one case, it's 
> CentOS 5.11 running OpenSIPs 1.8.1, the other (broken) is CentOS 6.7 running 
> OpenSIPs 1.11.5.
>   
> Call Flow in Trace:
> 10.10.10.205 10.10.8.103
>--Invite-->Indialog reinvite, so SDP
> <--Trying--
> <--200 OK--With SDP
> --ACK -->  With SDP in 1.8.1, no SDP in 1.11.5
>   
> OpenSIPS 1.8.1:
> ---
> INVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> From: ;tag=B2B.483.37
> CSeq: 2 INVITE
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Contact: 
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: ;tag=B2B.483.37
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> CSeq: 2 INVITE
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: ;tag=B2B.483.37
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> CSeq: 2 INVITE
> Contact: 
> Content-Type: application/sdp
> Content-Length:   185
>
> v=0
> o=VOIPSIL_SIP 416521046 416521046 IN IP4 10.10.8.103 s=Sip Call c=IN
> IP4 10.10.8.33
> t=0 0
> m=audio 21260 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
>
> ACK sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.de759b97.0
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> From: ;tag=B2B.483.37
> CSeq: 2 ACK
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.1

Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5

2016-03-24 Thread Louis Rochon

I recompiled with the 1.11.6, and did not resolve the issue. Traces show 
behaviour identical between 1.11.5 and 1.11.6.

Looking at the release notes for 1.11.6, there are some fixes that are closely 
related, but don't fix my specific issue. i.e. 95f5f79, 26a0a62. 

Any suggestions? Is there some yet-to-be documented B2BLogic module parameter 
that needs to be set? 

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
Sent: Monday, March 21, 2016 2:22 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

Thanks Liviu,

I will recompile and give it a try.

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Monday, March 21, 2016 9:31 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

Hi Louis,

There was a similar discussion back in August [1], concluding with a fix [2], 
made between the 1.11.5 and 1.11.6 releases. Updating to the latest OpenSIPS 
1.11 will most likely solve this problem.

[1]: http://lists.opensips.org/pipermail/users/2015-August/032239.html
[2]: https://github.com/OpenSIPS/opensips/commit/95f5f79b9250

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 21.03.2016 14:55, Louis Rochon wrote:
> Anybody?
>
> Louis
>
>
> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
> Sent: Wednesday, March 16, 2016 10:02 AM
> To: users@lists.opensips.org
> Subject: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in
> 1.11.5
>
> SDP in ACK lost in OpenSIPS 1.11.5
>
> This is something that used to work in OpenSIPS 1.8.1, but seems to have been 
> broken 1.11.5.
>
> Using the B2BUA facilities, I make the second leg of the call using 
> b2b_init_request. Then, if required, I move the call to another user agent 
> via a bash shell script, which issues a opensipsctl fifo b2b_bridge.
>
> For this in-dialog reinvite to work, there has to be an SDP in the ACK, which 
> is missing the trace produced using OpenSIPS 1.11.5. It's there in 1.8.1.
>
> Anyway of reintroducing the SDP in the ACK in 1.11.5?
>
> Trace Explanation
> -
> Objects: 10.10.8.103: source of initial call. Recipipient of reinvite.
>   10.10.10.205: CentOS running OpenSIPS with B2BUA. In one case, it's 
> CentOS 5.11 running OpenSIPs 1.8.1, the other (broken) is CentOS 6.7 running 
> OpenSIPs 1.11.5.
>   
> Call Flow in Trace:
> 10.10.10.205 10.10.8.103
>--Invite-->Indialog reinvite, so SDP
> <--Trying--
> <--200 OK--With SDP
> --ACK -->  With SDP in 1.8.1, no SDP in 1.11.5
>   
> OpenSIPS 1.8.1:
> ---
> INVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> From: ;tag=B2B.483.37
> CSeq: 2 INVITE
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Contact: 
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: ;tag=B2B.483.37
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> CSeq: 2 INVITE
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: ;tag=B2B.483.37
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> CSeq: 2 INVITE
> Contact: 
> Content-Type: application/sdp
> Content-Length:   185
>
> v=0
> o=VOIPSIL_SIP 416521046 416521046 IN IP4 10.10.8.103 s=Sip Call c=IN
> IP4 10.10.8.33
> t=0 0
> m=audio 21260 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
>
> ACK sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.de759b97.0
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> From: ;tag=B2B.483.37
> CSeq: 2 ACK
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> Max-Forwards: 70
> Content-Length: 186
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Allow: INVITE, ACK, CANCEL, BYE
> Priority: emergency
> Calluid: 1537A73F8B0C005C
> Geolocation: 
> Geolocation-Routing: yes
> Initial-CallID: 113c305d-89188638-113c3058-a664e63f@10.1

Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5

2016-03-21 Thread Louis Rochon

Thanks Liviu,

I will recompile and give it a try.

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Monday, March 21, 2016 9:31 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

Hi Louis,

There was a similar discussion back in August [1], concluding with a fix [2], 
made between the 1.11.5 and 1.11.6 releases. Updating to the latest OpenSIPS 
1.11 will most likely solve this problem.

[1]: http://lists.opensips.org/pipermail/users/2015-August/032239.html
[2]: https://github.com/OpenSIPS/opensips/commit/95f5f79b9250

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 21.03.2016 14:55, Louis Rochon wrote:
> Anybody?
>
> Louis
>
>
> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
> Sent: Wednesday, March 16, 2016 10:02 AM
> To: users@lists.opensips.org
> Subject: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
> 1.11.5
>
> SDP in ACK lost in OpenSIPS 1.11.5
>
> This is something that used to work in OpenSIPS 1.8.1, but seems to have been 
> broken 1.11.5.
>
> Using the B2BUA facilities, I make the second leg of the call using 
> b2b_init_request. Then, if required, I move the call to another user agent 
> via a bash shell script, which issues a opensipsctl fifo b2b_bridge.
>
> For this in-dialog reinvite to work, there has to be an SDP in the ACK, which 
> is missing the trace produced using OpenSIPS 1.11.5. It's there in 1.8.1.
>
> Anyway of reintroducing the SDP in the ACK in 1.11.5?
>
> Trace Explanation
> -
> Objects: 10.10.8.103: source of initial call. Recipipient of reinvite.
>   10.10.10.205: CentOS running OpenSIPS with B2BUA. In one case, it's 
> CentOS 5.11 running OpenSIPs 1.8.1, the other (broken) is CentOS 6.7 running 
> OpenSIPs 1.11.5.
>   
> Call Flow in Trace:
> 10.10.10.205 10.10.8.103
>--Invite-->Indialog reinvite, so SDP
> <--Trying--
> <--200 OK--With SDP
> --ACK -->  With SDP in 1.8.1, no SDP in 1.11.5
>   
> OpenSIPS 1.8.1:
> ---
> INVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> From: ;tag=B2B.483.37
> CSeq: 2 INVITE
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Contact: 
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: ;tag=B2B.483.37
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> CSeq: 2 INVITE
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: ;tag=B2B.483.37
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> CSeq: 2 INVITE
> Contact: 
> Content-Type: application/sdp
> Content-Length:   185
>
> v=0
> o=VOIPSIL_SIP 416521046 416521046 IN IP4 10.10.8.103 s=Sip Call c=IN 
> IP4 10.10.8.33
> t=0 0
> m=audio 21260 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
>
> ACK sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.de759b97.0
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> From: ;tag=B2B.483.37
> CSeq: 2 ACK
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> Max-Forwards: 70
> Content-Length: 186
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Allow: INVITE, ACK, CANCEL, BYE
> Priority: emergency
> Calluid: 1537A73F8B0C005C
> Geolocation: 
> Geolocation-Routing: yes
> Initial-CallID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> Contact: 
> Content-Type: application/sdp
>
> v=0
> o=VOIPSIL_SIP 322321996 322321996 IN IP4 10.10.10.203 s=Sip Call c=IN 
> IP4 10.10.10.7
> t=0 0
> m=audio 21336 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> BYE sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK4037.cd227b95.0
> To: "4507710011" 
> ;tag=89188638-3ac4d92f-ver59
> From: ;tag=B2B.87.164
> CSeq: 3 BYE
> Call-ID: ae3179a8-89188638-ae3179a3-3ac4d92f@10.10.8.103
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux)

Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5

2016-03-21 Thread Liviu Chircu


Hi Louis,

There was a similar discussion back in August [1], concluding with a fix 
[2], made between the 1.11.5 and 1.11.6 releases. Updating to the latest 
OpenSIPS 1.11 will most likely solve this problem.


[1]: http://lists.opensips.org/pipermail/users/2015-August/032239.html
[2]: https://github.com/OpenSIPS/opensips/commit/95f5f79b9250

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 21.03.2016 14:55, Louis Rochon wrote:

Anybody?

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
Sent: Wednesday, March 16, 2016 10:02 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5

SDP in ACK lost in OpenSIPS 1.11.5

This is something that used to work in OpenSIPS 1.8.1, but seems to have been 
broken 1.11.5.

Using the B2BUA facilities, I make the second leg of the call using 
b2b_init_request. Then, if required, I move the call to another user agent via 
a bash shell script, which issues a opensipsctl fifo b2b_bridge.

For this in-dialog reinvite to work, there has to be an SDP in the ACK, which 
is missing the trace produced using OpenSIPS 1.11.5. It's there in 1.8.1.

Anyway of reintroducing the SDP in the ACK in 1.11.5?

Trace Explanation
-
Objects: 10.10.8.103: source of initial call. Recipipient of reinvite.
  10.10.10.205: CentOS running OpenSIPS with B2BUA. In one case, it's 
CentOS 5.11 running OpenSIPs 1.8.1, the other (broken) is CentOS 6.7 running 
OpenSIPs 1.11.5.

Call Flow in Trace:
10.10.10.205 10.10.8.103
   --Invite-->Indialog reinvite, so SDP
  <--Trying--
  <--200 OK--With SDP
  --ACK -->  With SDP in 1.8.1, no SDP in 1.11.5

OpenSIPS 1.8.1:
---
INVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
From: ;tag=B2B.483.37
CSeq: 2 INVITE
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
Contact: 

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
From: ;tag=B2B.483.37
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
CSeq: 2 INVITE
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
From: ;tag=B2B.483.37
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
CSeq: 2 INVITE
Contact: 
Content-Type: application/sdp
Content-Length:   185

v=0
o=VOIPSIL_SIP 416521046 416521046 IN IP4 10.10.8.103 s=Sip Call c=IN IP4 
10.10.8.33
t=0 0
m=audio 21260 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

ACK sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.de759b97.0
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
From: ;tag=B2B.483.37
CSeq: 2 ACK
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
Max-Forwards: 70
Content-Length: 186
User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
Allow: INVITE, ACK, CANCEL, BYE
Priority: emergency
Calluid: 1537A73F8B0C005C
Geolocation: 
Geolocation-Routing: yes
Initial-CallID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
Contact: 
Content-Type: application/sdp

v=0
o=VOIPSIL_SIP 322321996 322321996 IN IP4 10.10.10.203 s=Sip Call c=IN IP4 
10.10.10.7
t=0 0
m=audio 21336 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
BYE sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK4037.cd227b95.0
To: "4507710011" 
;tag=89188638-3ac4d92f-ver59
From: ;tag=B2B.87.164
CSeq: 3 BYE
Call-ID: ae3179a8-89188638-ae3179a3-3ac4d92f@10.10.8.103
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
Contact: 


OpenSIPS 1.11.5:

NVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
To: "4507710011" 
;tag=89188638-1dace611-ver5b
From: ;tag=B2B.11.465
CSeq: 2 INVITE
Call-ID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS (1.11.5-notls (x86_64/linux))
Contact: 

SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
From: ;tag=B2B.11.465
To: "4507710011" 
;tag=89188638-1dace611-ver5b
Call-ID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
CSeq: 2 INVITE
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
From: ;tag=B2B.11.465
To: "4507710011" 
;tag=89188638-1dace611-ver5b
Call-ID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
CSeq: 2 INVITE
Contact: 
Content-T

Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5

2016-03-21 Thread Louis Rochon

Anybody?

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
Sent: Wednesday, March 16, 2016 10:02 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5

SDP in ACK lost in OpenSIPS 1.11.5

This is something that used to work in OpenSIPS 1.8.1, but seems to have been 
broken 1.11.5.

Using the B2BUA facilities, I make the second leg of the call using 
b2b_init_request. Then, if required, I move the call to another user agent via 
a bash shell script, which issues a opensipsctl fifo b2b_bridge.

For this in-dialog reinvite to work, there has to be an SDP in the ACK, which 
is missing the trace produced using OpenSIPS 1.11.5. It's there in 1.8.1.

Anyway of reintroducing the SDP in the ACK in 1.11.5?

Trace Explanation
-
Objects: 10.10.8.103: source of initial call. Recipipient of reinvite.
 10.10.10.205: CentOS running OpenSIPS with B2BUA. In one case, it's 
CentOS 5.11 running OpenSIPs 1.8.1, the other (broken) is CentOS 6.7 running 
OpenSIPs 1.11.5.
 
Call Flow in Trace:
10.10.10.205 10.10.8.103
  --Invite-->Indialog reinvite, so SDP
  <--Trying--
  <--200 OK--With SDP
  --ACK -->  With SDP in 1.8.1, no SDP in 1.11.5
  
OpenSIPS 1.8.1:
---
INVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
From: ;tag=B2B.483.37
CSeq: 2 INVITE
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
Contact: 

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
From: ;tag=B2B.483.37
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
CSeq: 2 INVITE
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
From: ;tag=B2B.483.37
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
CSeq: 2 INVITE
Contact: 
Content-Type: application/sdp
Content-Length:   185

v=0
o=VOIPSIL_SIP 416521046 416521046 IN IP4 10.10.8.103 s=Sip Call c=IN IP4 
10.10.8.33
t=0 0
m=audio 21260 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

ACK sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.de759b97.0
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
From: ;tag=B2B.483.37
CSeq: 2 ACK
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
Max-Forwards: 70
Content-Length: 186
User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
Allow: INVITE, ACK, CANCEL, BYE
Priority: emergency
Calluid: 1537A73F8B0C005C
Geolocation: 
Geolocation-Routing: yes
Initial-CallID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
Contact: 
Content-Type: application/sdp

v=0
o=VOIPSIL_SIP 322321996 322321996 IN IP4 10.10.10.203 s=Sip Call c=IN IP4 
10.10.10.7
t=0 0
m=audio 21336 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
BYE sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK4037.cd227b95.0
To: "4507710011" 
;tag=89188638-3ac4d92f-ver59
From: ;tag=B2B.87.164
CSeq: 3 BYE
Call-ID: ae3179a8-89188638-ae3179a3-3ac4d92f@10.10.8.103
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
Contact: 


OpenSIPS 1.11.5:

NVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
To: "4507710011" 
;tag=89188638-1dace611-ver5b
From: ;tag=B2B.11.465
CSeq: 2 INVITE
Call-ID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS (1.11.5-notls (x86_64/linux))
Contact: 

SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
From: ;tag=B2B.11.465
To: "4507710011" 
;tag=89188638-1dace611-ver5b
Call-ID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
CSeq: 2 INVITE
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
From: ;tag=B2B.11.465
To: "4507710011" 
;tag=89188638-1dace611-ver5b
Call-ID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
CSeq: 2 INVITE
Contact: 
Content-Type: application/sdp
Content-Length:   185

v=0
o=VOIPSIL_SIP 415919281 415919281 IN IP4 10.10.8.103 s=Sip Call c=IN IP4 
10.10.8.33
t=0 0
m=audio 21260 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

ACK sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.4d149882.0
To: "4507710011" 
;tag=89188638-1dace611-ver5b
From: ;tag=B2B.11.465
CSeq: 2 ACK
Call-ID: 76579f54-8918

[OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5

2016-03-19 Thread Louis Rochon

SDP in ACK lost in OpenSIPS 1.11.5

This is something that used to work in OpenSIPS 1.8.1, but seems to have been 
broken 1.11.5.

Using the B2BUA facilities, I make the second leg of the call using 
b2b_init_request. Then, if required, I move the call to another user agent via 
a bash shell script, which issues a opensipsctl fifo b2b_bridge.

For this in-dialog reinvite to work, there has to be an SDP in the ACK, which 
is missing the trace produced using OpenSIPS 1.11.5. It's there in 1.8.1.

Anyway of reintroducing the SDP in the ACK in 1.11.5?

Trace Explanation
-
Objects: 10.10.8.103: source of initial call. Recipipient of reinvite.
 10.10.10.205: CentOS running OpenSIPS with B2BUA. In one case, it's 
CentOS 5.11 running OpenSIPs 1.8.1, the other (broken) is CentOS 6.7 running 
OpenSIPs 1.11.5.
 
Call Flow in Trace:
10.10.10.205 10.10.8.103
  --Invite-->Indialog reinvite, so SDP
  <--Trying--
  <--200 OK--With SDP
  --ACK -->  With SDP in 1.8.1, no SDP in 1.11.5
  
OpenSIPS 1.8.1:
---
INVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
From: ;tag=B2B.483.37
CSeq: 2 INVITE
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
Contact: 

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
From: ;tag=B2B.483.37
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
CSeq: 2 INVITE
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
From: ;tag=B2B.483.37
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
CSeq: 2 INVITE
Contact: 
Content-Type: application/sdp
Content-Length:   185

v=0
o=VOIPSIL_SIP 416521046 416521046 IN IP4 10.10.8.103
s=Sip Call
c=IN IP4 10.10.8.33
t=0 0
m=audio 21260 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

ACK sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.de759b97.0
To: "4507710011" 
;tag=89188638-a664e63f-ver5d
From: ;tag=B2B.483.37
CSeq: 2 ACK
Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
Max-Forwards: 70
Content-Length: 186
User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
Allow: INVITE, ACK, CANCEL, BYE
Priority: emergency
Calluid: 1537A73F8B0C005C
Geolocation: 
Geolocation-Routing: yes
Initial-CallID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
Contact: 
Content-Type: application/sdp

v=0
o=VOIPSIL_SIP 322321996 322321996 IN IP4 10.10.10.203
s=Sip Call
c=IN IP4 10.10.10.7
t=0 0
m=audio 21336 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
BYE sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK4037.cd227b95.0
To: "4507710011" 
;tag=89188638-3ac4d92f-ver59
From: ;tag=B2B.87.164
CSeq: 3 BYE
Call-ID: ae3179a8-89188638-ae3179a3-3ac4d92f@10.10.8.103
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
Contact: 


OpenSIPS 1.11.5:

NVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
To: "4507710011" 
;tag=89188638-1dace611-ver5b
From: ;tag=B2B.11.465
CSeq: 2 INVITE
Call-ID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS (1.11.5-notls (x86_64/linux))
Contact: 

SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
From: ;tag=B2B.11.465
To: "4507710011" 
;tag=89188638-1dace611-ver5b
Call-ID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
CSeq: 2 INVITE
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.3d149882.0
From: ;tag=B2B.11.465
To: "4507710011" 
;tag=89188638-1dace611-ver5b
Call-ID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
CSeq: 2 INVITE
Contact: 
Content-Type: application/sdp
Content-Length:   185

v=0
o=VOIPSIL_SIP 415919281 415919281 IN IP4 10.10.8.103
s=Sip Call
c=IN IP4 10.10.8.33
t=0 0
m=audio 21260 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

ACK sip:4507710011@10.10.8.103;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.205:5060;branch=z9hG4bK97f1.4d149882.0
To: "4507710011" 
;tag=89188638-1dace611-ver5b
From: ;tag=B2B.11.465
CSeq: 2 ACK
Call-ID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS (1.11.5-notls (x86_64/linux))
Allow: INVITE, ACK, CANCEL, BYE
Priority: emergency
Calluid: 1537A68E4C0C005A
Geolocation: 
Geolocation-Routing: yes
Initial-CallID: 76579f54-89188638-76579f4f-1dace611@10.10.8.103
Contact: 
Content-Type: application/sdp

Louis Rochon
Senior Systems Designer
Solacom Technologies

_

Re: [OpenSIPS-Users] sdp answer

2016-01-12 Thread Bogdan-Andrei Iancu


Hi Riko,

I understand, but my knowledge over rtpengine is not so good. Maybe you 
should ask over the rptengine project.


Best regards,

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

On 12.01.2016 08:24, riko nir wrote:

Hi Andrei,

The opensips and rtpengine is interacting, for overwriting the SDP. We 
tested "JsSIP client with (opesips + rtpengine)" and it is working fine.


Here, DTLS handshake is happening using rtpengine, and so SRTP is 
handling by rtpengine.


But I don't want rtpengine to handle SRTP traffic. Only the "DTLS 
handshake" needs to be handled by RTPEngine and I just need the crypto 
method and the crypto keys from rtpengine to be sent to opensips.

Do you have any suggestion for fetching the crypto keys from rtpengine?

Thanks,
Riko

On Mon, Jan 11, 2016 at 4:24 PM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


When OpenSIPS handles the SIP messages carrying SDP (typically
INVITE request and 200 OK reply), OpenSIPS will communicate with
rtpengine and update the SDP with the new IP and port. Never
tested, by AFAIK rtpengine may know to handle SRTP...not sure, you
may check with the project.

Regards,

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

On 11.01.2016 12:00, riko nir wrote:

Ok. Incase if the media needs to go via rtpengine, then how the
signaling happens, and how the rtpengine is forwarding the media
to the other end?

In case of DTLS, only the DTLS handlshake needs to be taken care
by the RtpEngine, and the SRTP is handled by some other media
server, and only the DTLS-key needs to be send to the media
server. In that case, how can we fetch dtls-key after the DTLS
handshake?

Thanks.

On Mon, Jan 11, 2016 at 3:13 PM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Riko,

There are no SDP answers created by opensips - it simply
changes the received SDP offer and it forwards it to the next
destination - basically it is doing proxying (performing some
changes too) of the SDP offers between the end points.

Best regards,

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

On 06.01.2016 15:42, riko nir wrote:

Hi all,

How is the communication flow happens between opensips and
rtpengine incase of a call through SIP over websocket?
When opensips receives a sdp offer from a web-based sip
client, opensips is creating sdp answer or rtpengine is
creating sdp anwer or rtpengine just updating ice
information only in sdp and other than that, media related
info is created by opensips?

Thanks.
Riko


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Re: [OpenSIPS-Users] sdp answer

2016-01-11 Thread Bogdan-Andrei Iancu

When OpenSIPS handles the SIP messages carrying SDP (typically INVITE 
request and 200 OK reply), OpenSIPS will communicate with rtpengine and 
update the SDP with the new IP and port. Never tested, by AFAIK 
rtpengine may know to handle SRTP...not sure, you may check with the 
project.


Regards,

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

On 11.01.2016 12:00, riko nir wrote:
Ok. Incase if the media needs to go via rtpengine, then how the 
signaling happens, and how the rtpengine is forwarding the media to 
the other end?


In case of DTLS, only the DTLS handlshake needs to be taken care by 
the RtpEngine, and the SRTP is handled by some other media server, and 
only the DTLS-key needs to be send to the media server. In that case, 
how can we fetch dtls-key after the DTLS handshake?


Thanks.

On Mon, Jan 11, 2016 at 3:13 PM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Riko,

There are no SDP answers created by opensips - it simply changes
the received SDP offer and it forwards it to the next destination
- basically it is doing proxying (performing some changes too) of
the SDP offers between the end points.

Best regards,

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

On 06.01.2016 15:42, riko nir wrote:

Hi all,

How is the communication flow happens between opensips and
rtpengine incase of a call through SIP over websocket?
When opensips receives a sdp offer from a web-based sip client,
opensips is creating sdp answer or rtpengine is creating sdp
anwer or rtpengine just updating ice information only in sdp and
other than that, media related info is created by opensips?

Thanks.
Riko


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Re: [OpenSIPS-Users] sdp answer

2016-01-11 Thread riko nir

Ok. Incase if the media needs to go via rtpengine, then how the signaling
happens, and how the rtpengine is forwarding the media to the other end?

In case of DTLS, only the DTLS handlshake needs to be taken care by the
RtpEngine, and the SRTP is handled by some other media server, and only the
DTLS-key needs to be send to the media server. In that case, how can we
fetch dtls-key after the DTLS handshake?

Thanks.

On Mon, Jan 11, 2016 at 3:13 PM, Bogdan-Andrei Iancu 
wrote:

> Hi Riko,
>
> There are no SDP answers created by opensips - it simply changes the
> received SDP offer and it forwards it to the next destination - basically
> it is doing proxying (performing some changes too) of the SDP offers
> between the end points.
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 06.01.2016 15:42, riko nir wrote:
>
> Hi all,
>
> How is the communication flow happens between opensips and rtpengine
> incase of a call through SIP over websocket?
> When opensips receives a sdp offer from a web-based sip client, opensips
> is creating sdp answer or rtpengine is creating sdp anwer or rtpengine just
> updating ice information only in sdp and other than that, media related
> info is created by opensips?
>
> Thanks.
> Riko
>
>
> ___
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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Re: [OpenSIPS-Users] sdp answer

2016-01-11 Thread Bogdan-Andrei Iancu


Hi Riko,

There are no SDP answers created by opensips - it simply changes the 
received SDP offer and it forwards it to the next destination - 
basically it is doing proxying (performing some changes too) of the SDP 
offers between the end points.


Best regards,

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

On 06.01.2016 15:42, riko nir wrote:

Hi all,

How is the communication flow happens between opensips and rtpengine 
incase of a call through SIP over websocket?
When opensips receives a sdp offer from a web-based sip client, 
opensips is creating sdp answer or rtpengine is creating sdp anwer or 
rtpengine just updating ice information only in sdp and other than 
that, media related info is created by opensips?


Thanks.
Riko


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[OpenSIPS-Users] sdp answer

2016-01-06 Thread riko nir

Hi all,

How is the communication flow happens between opensips and rtpengine incase
of a call through SIP over websocket?
When opensips receives a sdp offer from a web-based sip client, opensips is
creating sdp answer or rtpengine is creating sdp anwer or rtpengine just
updating ice information only in sdp and other than that, media related
info is created by opensips?

Thanks.
Riko
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Re: [OpenSIPS-Users] SDP edit

2014-06-19 Thread Saúl Ibarra Corretgé

Checkout the textops module: 
http://www.opensips.org/html/docs/modules/devel/textops.html

On Jun 19, 2014, at 10:47 PM, Ionut Muntean wrote:

> Hello,
> 
> Is there a easy way to remove some attributes present in the SDP (b=RS:xx 
> and/or b=RR:xx)?
> 
> Thank you.
> 
> / Ionut Muntean
> 
> ___
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--
Saúl Ibarra Corretgé
AG Projects




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[OpenSIPS-Users] SDP edit

2014-06-19 Thread Ionut Muntean


Hello,

Is there a easy way to remove some attributes present in the SDP 
(b=RS:xx and/or b=RR:xx)?


Thank you.

/ Ionut Muntean

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Re: [OpenSIPS-Users] SDP change issue

2014-04-14 Thread Răzvan Crainea


Hi, Chen-Che!

No, OpenSIPS does not have any built-in mechanism for this. As far as I 
understand, you sometimes need to rewrite the IP OpenSIPS advertises in 
the SDP. You can specify the new IP in the second parameter of the 
rtpproxy_offer/answer() function.


Best regards,

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

On 04/14/2014 09:49 AM, microx wrote:

Dear all,

I'm encountering an issue as follows. In the system, there are one SIP
server (OpenSIPS) and multiple RTP proxies. Each RTP proxy serves for one
region and the users in that region will use the corresponding RTP proxy for
media streaming relay. To achieve this, I make the users in the same region
have the same prefix in their SIP numbers and create a mapping between the
prefix and the RTP proxy. With such setup, users will be served by a close
RTP proxy and the latency would be reduced. By setting the RTP proxy set
(id) to the corresponding prefix, such a scenario can be realized. However,
in the system, the RTP proxies listen on private interfaces for security
concern (an RTP proxy and some other applications run on a host behind NAT).
Thus, I need to rewrite the SDP connection IP with the corresponding NAT's
public IP for the RTP proxy. To do this, it seems somewhat complicated to do
configuration. I would like to know whether OpenSIPS has any built-in
mechanism for this issue (prefix + RTP proxy + NAT public IP). Many thanks
for any comment.

Best wishes,
Chen-Che



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[OpenSIPS-Users] SDP change issue

2014-04-13 Thread microx

Dear all,

I'm encountering an issue as follows. In the system, there are one SIP
server (OpenSIPS) and multiple RTP proxies. Each RTP proxy serves for one
region and the users in that region will use the corresponding RTP proxy for
media streaming relay. To achieve this, I make the users in the same region
have the same prefix in their SIP numbers and create a mapping between the
prefix and the RTP proxy. With such setup, users will be served by a close
RTP proxy and the latency would be reduced. By setting the RTP proxy set
(id) to the corresponding prefix, such a scenario can be realized. However,
in the system, the RTP proxies listen on private interfaces for security
concern (an RTP proxy and some other applications run on a host behind NAT).
Thus, I need to rewrite the SDP connection IP with the corresponding NAT's
public IP for the RTP proxy. To do this, it seems somewhat complicated to do
configuration. I would like to know whether OpenSIPS has any built-in
mechanism for this issue (prefix + RTP proxy + NAT public IP). Many thanks
for any comment.

Best wishes,
Chen-Che



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http://opensips-open-sip-server.1449251.n2.nabble.com/SDP-change-issue-tp7590721.html
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Re: [OpenSIPS-Users] SDP - doble

2012-04-16 Thread Saúl Ibarra Corretgé

Hi,

On Apr 16, 2012, at 10:34 AM, goup2010 wrote:

> Hello,
> 
> I test opensips 1.8.
> 
> When I use rtpproxy_offer()  and rtpproxy_answer() in SDP ,  line
> appears two times.
> 
> Is this correct?
> 

Yes. The one on the stream takes precedence. As per RFC4566, sec5.7:

A session description MUST contain either at least one "c=" field in
   each media description or a single "c=" field at the session level.
   It MAY contain a single session-level "c=" field and additional "c="
   field(s) per media description, in which case the per-media values
   override the session-level settings for the respective media.

Regards,

--
Saúl Ibarra Corretgé
AG Projects

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[OpenSIPS-Users] SDP - doble

2012-04-16 Thread goup2010

Hello,

I test opensips 1.8.

When I use rtpproxy_offer()  and rtpproxy_answer() in SDP ,  line
appears two times.

Is this correct?

Here is SDP

v=0
o=- 3543552561 3543552561 IN IP4 192.168.1.102
s=pjmedia
c=IN IP4 192.168.1.102
b=AS:84
t=0 0
a=X-nat:0
m=audio 55946 RTP/AVP 8 0 101
c=IN IP4 88.99.100.100
b=TIAS:64000
a=rtcp:55947 IN IP4 88.99.100.100
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Best regards,
PlayMen

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Re: [OpenSIPS-Users] SDP session inspection in OpenSIPs

2009-09-02 Thread urmi lakkad

Hello,

Thank you very much Bogdan and Ovidiu for quick response.

I have used dialog  'dlg_list_ctx' mi command. And I can inspect the dialog
as well as the associated qos session.

-Urmi

On Tue, Sep 1, 2009 at 7:49 PM, Ovidiu Sas  wrote:

> Hello Urmi,
>
> Use the dialog 'dlg_list_ctx' mi command to inspect the internals of
> the dialog along with the
> associated qos session:
> http://www.opensips.org/html/docs/modules/1.5.x/dialog.html#id272808
> This is stated in the qos module doc:
> http://www.opensips.org/html/docs/modules/1.5.x/qos.html#id227316
>
> Regards,
> Ovidiu Sas
>
> On Tue, Sep 1, 2009 at 9:29 AM, urmi lakkad wrote:
> > Hello,
> >
> > I am using Opensips-1.5.1 with Asterisk.
> > I want to test the QOS module functionality. I have configured the dialog
> > module and its working fine.
> > Even I have set the flag of QOS module. Now my question is how to inspect
> > the SDP session.
> >
> > Thanks for your attention.
> >
> > -Urmi
> >
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> >
>
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Re: [OpenSIPS-Users] SDP session inspection in OpenSIPs

2009-09-01 Thread Ovidiu Sas

Hello Urmi,

Use the dialog 'dlg_list_ctx' mi command to inspect the internals of
the dialog along with the
associated qos session:
http://www.opensips.org/html/docs/modules/1.5.x/dialog.html#id272808
This is stated in the qos module doc:
http://www.opensips.org/html/docs/modules/1.5.x/qos.html#id227316

Regards,
Ovidiu Sas

On Tue, Sep 1, 2009 at 9:29 AM, urmi lakkad wrote:
> Hello,
>
> I am using Opensips-1.5.1 with Asterisk.
> I want to test the QOS module functionality. I have configured the dialog
> module and its working fine.
> Even I have set the flag of QOS module. Now my question is how to inspect
> the SDP session.
>
> Thanks for your attention.
>
> -Urmi
>
> ___
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> Users@lists.opensips.org
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>

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Re: [OpenSIPS-Users] SDP session inspection in OpenSIPs

2009-09-01 Thread Bogdan-Andrei Iancu

Hi Urmi,

There are some functions in textops module to allow you codec inspection 
and manipulation:
http://www.opensips.org/Main/News0034

Regards,
Bogdan


urmi lakkad wrote:
>
> Hello,
>
> I am using Opensips-1.5.1 with Asterisk.
> I want to test the QOS module functionality. I have configured the 
> dialog module and its working fine.
> Even I have set the flag of QOS module. Now my question is how to 
> inspect the SDP session.
>
> Thanks for your attention.
>
> -Urmi
>
> 
>
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>   


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[OpenSIPS-Users] SDP session inspection in OpenSIPs

2009-09-01 Thread urmi lakkad

Hello,

 I am using Opensips-1.5.1 with Asterisk.
I want to test the QOS module functionality. I have configured the dialog
module and its working fine.
Even I have set the flag of QOS module. Now my question is how to inspect
the SDP session.

 Thanks for your attention.

 -Urmi
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