date:20190731

Re: [OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 and 2.4.2

2019-07-31 Thread Louis Rochon

I have gone over the release notes for 2.4.2 to 2.4.6. I will upgrade and 
retest.

Louis

From: Users  On Behalf Of Ben Newlin
Sent: July 31, 2019 10:34 AM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 
1.8 and 2.4.2

WARNING: External Email: Exercise Caution

I’m not sure about the B2B behavior, but is there a reason you are not 
migrating to 2.4.6, which is the latest supported 2.4 release? It’s possible 
this problem has already been reported and fixed.

Ben Newlin

From: Users 
mailto:users-boun...@lists.opensips.org>> on 
behalf of Louis Rochon 
mailto:louis.roc...@comtechtel.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Wednesday, July 31, 2019 at 10:29 AM
To: "users@lists.opensips.org" 
mailto:users@lists.opensips.org>>
Subject: [OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 
and 2.4.2

Trying to upgrade my platform to OpenSIPS 2.4.2 (yea, about time, I know).

We use this to move established calls from one server to another:

/usr/local/sbin/opensipsctl fifo b2b_bridge  sip:555@1.2.3.4 0

This works fine with OpenSIPS 1.8. The above causes this:

1. Invite (reinvite) to caller with no sdp.

2. RX of 200 OK with SDP from caller

3. Invite with SDP to 1.2.3.4

4. RX of 200 OK with SDP from 1.2.3.4

5. OpenSIPS send ACK WITH SDP to caller.

Call moved. All good.

With OpenSIPS 2.4.2:

1. Invite (reinvite) to caller with no sdp.

2. RX of 200 OK with SDP from caller

3. Invite with SDP to 1.2.3.4

4. RX of 200 OK with SDP from 1.2.3.4

5. OpenSIPS send ACK WITHOUT SDP to caller.

With 2.4.2, I get one way audio, very likely because of the lack of SDP in the 
ACK in point 5.

My research does not lead me anywhere on how why this behaviour changed between 
1.8 and 2.4.2. I can't find any flags to set to get b2b_bridge command to have 
to old behaviour.

Help!

regards,

Louis
NOTICE TO RECIPIENT: This email, including attachments, may contain information 
which is confidential, proprietary, attorney-client privileged and / or 
controlled under U.S. export laws and regulations and may be restricted from 
disclosure by applicable State and Federal law. Nothing in this email shall 
create any legal binding agreement between the parties unless expressly stated 
herein and provided by an authorized representative of Comtech 
Telecommunications Corp. or its subsidiaries. If you are not the intended 
recipient of this message, be advised that any dissemination, distribution, or 
use of the contents of this message is strictly prohibited. If you received 
this message in error, please notify us immediately by return email and 
permanently delete all copies of the original email and any attached 
documentation from any computer or other media.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 and 2.4.2

2019-07-31 Thread Ben Newlin

I’m not sure about the B2B behavior, but is there a reason you are not 
migrating to 2.4.6, which is the latest supported 2.4 release? It’s possible 
this problem has already been reported and fixed.

Ben Newlin

From: Users  on behalf of Louis Rochon 

Reply-To: OpenSIPS users mailling list 
Date: Wednesday, July 31, 2019 at 10:29 AM
To: "users@lists.opensips.org" 
Subject: [OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 
and 2.4.2

Trying to upgrade my platform to OpenSIPS 2.4.2 (yea, about time, I know).

We use this to move established calls from one server to another:

/usr/local/sbin/opensipsctl fifo b2b_bridge  sip:555@1.2.3.4 0

This works fine with OpenSIPS 1.8. The above causes this:

1. Invite (reinvite) to caller with no sdp.

2. RX of 200 OK with SDP from caller

3. Invite with SDP to 1.2.3.4

4. RX of 200 OK with SDP from 1.2.3.4

5. OpenSIPS send ACK WITH SDP to caller.

Call moved. All good.

With OpenSIPS 2.4.2:

1. Invite (reinvite) to caller with no sdp.

2. RX of 200 OK with SDP from caller

3. Invite with SDP to 1.2.3.4

4. RX of 200 OK with SDP from 1.2.3.4

5. OpenSIPS send ACK WITHOUT SDP to caller.

With 2.4.2, I get one way audio, very likely because of the lack of SDP in the 
ACK in point 5.

My research does not lead me anywhere on how why this behaviour changed between 
1.8 and 2.4.2. I can't find any flags to set to get b2b_bridge command to have 
to old behaviour.

Help!

regards,

Louis
NOTICE TO RECIPIENT: This email, including attachments, may contain information 
which is confidential, proprietary, attorney-client privileged and / or 
controlled under U.S. export laws and regulations and may be restricted from 
disclosure by applicable State and Federal law. Nothing in this email shall 
create any legal binding agreement between the parties unless expressly stated 
herein and provided by an authorized representative of Comtech 
Telecommunications Corp. or its subsidiaries. If you are not the intended 
recipient of this message, be advised that any dissemination, distribution, or 
use of the contents of this message is strictly prohibited. If you received 
this message in error, please notify us immediately by return email and 
permanently delete all copies of the original email and any attached 
documentation from any computer or other media.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 and 2.4.2

2019-07-31 Thread Louis Rochon

Trying to upgrade my platform to OpenSIPS 2.4.2 (yea, about time, I know).



We use this to move established calls from one server to another:



/usr/local/sbin/opensipsctl fifo b2b_bridge  sip:555@1.2.3.4 0



This works fine with OpenSIPS 1.8. The above causes this:

1. Invite (reinvite) to caller with no sdp.

2. RX of 200 OK with SDP from caller

3. Invite with SDP to 1.2.3.4

4. RX of 200 OK with SDP from 1.2.3.4

5. OpenSIPS send ACK WITH SDP to caller.



Call moved. All good.



With OpenSIPS 2.4.2:

1. Invite (reinvite) to caller with no sdp.

2. RX of 200 OK with SDP from caller

3. Invite with SDP to 1.2.3.4

4. RX of 200 OK with SDP from 1.2.3.4

5. OpenSIPS send ACK WITHOUT SDP to caller.



With 2.4.2, I get one way audio, very likely because of the lack of SDP in the 
ACK in point 5.



My research does not lead me anywhere on how why this behaviour changed between 
1.8 and 2.4.2. I can't find any flags to set to get b2b_bridge command to have 
to old behaviour.



Help!



regards,

Louis

NOTICE TO RECIPIENT: This email, including attachments, may contain information 
which is confidential, proprietary, attorney-client privileged and / or 
controlled under U.S. export laws and regulations and may be restricted from 
disclosure by applicable State and Federal law. Nothing in this email shall 
create any legal binding agreement between the parties unless expressly stated 
herein and provided by an authorized representative of Comtech 
Telecommunications Corp. or its subsidiaries. If you are not the intended 
recipient of this message, be advised that any dissemination, distribution, or 
use of the contents of this message is strictly prohibited. If you received 
this message in error, please notify us immediately by return email and 
permanently delete all copies of the original email and any attached 
documentation from any computer or other media.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Johan De Clercq

And no need for ice=force. You can drop that. Also check your sdp settings.

On Wed, 31 Jul 2019, 15:15 David Villasmil, 
wrote:

> Hello,
>
> You need to do this for every leg of the call. This means:
>
> Call from SRTP client TO non-SRTP:
> Remove the ICE, etc.
>
> When the REPLY with the 200 SDP comes back FROM the non-SRTP, you need to
> ADD ICE, etc.
>
> Hope that makes sense
>
> David
>
> On Wed, 31 Jul 2019 at 14:03, Dragomir Haralambiev 
> wrote:
>
>> Hi,
>> When change the answer flag to
>>
>> $var(rtpengine_flags) = " RTP/SAVP  rtcp-mux-offer ICE=force";
>>  rtpengine_answer("$var(rtpengine_flags)");
>>
>> Call is connected but UAC1 not send and receive voices.
>>
>> Regards,
>>
>> Dragomir
>>
>> На ср, 31.07.2019 г. в 15:53 ч. Sasmita Panda 
>> написа:
>>
>>> Hi Dragomir,
>>>
>>> I had mentioned to modify this according to your requirement .   If your
>>> phone only support RTP/SAVP then change the flag what I have mentioned
>>> while answering .
>>>
>>>
>>> *Thanks & Regards*
>>> *Sasmita Panda*
>>> *Senior Network Testing and Software Engineer*
>>> *3CLogic , ph:07827611765*
>>>
>>>
>>> On Wed, Jul 31, 2019 at 6:17 PM Johan De Clercq 
>>> wrote:
>>>
 Use rtp/savp

 On Wed, 31 Jul 2019, 14:40 Dragomir Haralambiev, 
 wrote:

> Hi,
>
> Thanks for your replay, but this not working.
>
> UAC1 receive 183 session progress with:
> receive audio 50106 UDP/TLS/RTP/SAVP 0 8 18 101
>
> UAC1   send to Opensips CANCEL.
>
> I make test with MicroSips latest version.
>
> Best regards,
> Dragomir
>
> На ср, 31.07.2019 г. в 15:04 ч. Sasmita Panda 
> написа:
>
>> Hi ,
>>
>> You have to do something like below  wherever you are calling
>> rtpengine_offer/rtpengine_answer.
>>
>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>> replace-origin ICE=remove";
>>  rtpengine_offer("$var(rtpengine_flags)");
>>
>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVP rtcp-mux-offer ICE=force";
>>  rtpengine_answer("$var(rtpengine_flags)");
>>
>> You can modify this according to your requirement .
>>
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Senior Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>>
>>
>> On Wed, Jul 31, 2019 at 5:16 PM Dragomir Haralambiev <
>> goup2...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I have 2 applications connected to Opensips+rtpengine:
>>> UAC1 -use encryption always. SRTP (RTP/SAVP)
>>> UAC2 - never use encryption  . RTP (RTP/AVP)
>>>
>>> How to setup Opensips to make follow call:
>>> UAC1 SRTP -> Opensips+rtpengine ---> UAC2 RTP
>>>
>>> Thanks,
>>> Dragomir
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users

>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread David Villasmil

Hello,

You need to do this for every leg of the call. This means:

Call from SRTP client TO non-SRTP:
Remove the ICE, etc.

When the REPLY with the 200 SDP comes back FROM the non-SRTP, you need to
ADD ICE, etc.

Hope that makes sense

David

On Wed, 31 Jul 2019 at 14:03, Dragomir Haralambiev 
wrote:

> Hi,
> When change the answer flag to
>
> $var(rtpengine_flags) = " RTP/SAVP  rtcp-mux-offer ICE=force";
>  rtpengine_answer("$var(rtpengine_flags)");
>
> Call is connected but UAC1 not send and receive voices.
>
> Regards,
>
> Dragomir
>
> На ср, 31.07.2019 г. в 15:53 ч. Sasmita Panda  написа:
>
>> Hi Dragomir,
>>
>> I had mentioned to modify this according to your requirement .   If your
>> phone only support RTP/SAVP then change the flag what I have mentioned
>> while answering .
>>
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Senior Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>>
>>
>> On Wed, Jul 31, 2019 at 6:17 PM Johan De Clercq  wrote:
>>
>>> Use rtp/savp
>>>
>>> On Wed, 31 Jul 2019, 14:40 Dragomir Haralambiev, 
>>> wrote:
>>>
 Hi,

 Thanks for your replay, but this not working.

 UAC1 receive 183 session progress with:
 receive audio 50106 UDP/TLS/RTP/SAVP 0 8 18 101

 UAC1   send to Opensips CANCEL.

 I make test with MicroSips latest version.

 Best regards,
 Dragomir

 На ср, 31.07.2019 г. в 15:04 ч. Sasmita Panda 
 написа:

> Hi ,
>
> You have to do something like below  wherever you are calling
> rtpengine_offer/rtpengine_answer.
>
> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
> replace-origin ICE=remove";
>  rtpengine_offer("$var(rtpengine_flags)");
>
> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVP rtcp-mux-offer ICE=force";
>  rtpengine_answer("$var(rtpengine_flags)");
>
> You can modify this according to your requirement .
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
>
> On Wed, Jul 31, 2019 at 5:16 PM Dragomir Haralambiev <
> goup2...@gmail.com> wrote:
>
>> Hello,
>>
>> I have 2 applications connected to Opensips+rtpengine:
>> UAC1 -use encryption always. SRTP (RTP/SAVP)
>> UAC2 - never use encryption  . RTP (RTP/AVP)
>>
>> How to setup Opensips to make follow call:
>> UAC1 SRTP -> Opensips+rtpengine ---> UAC2 RTP
>>
>> Thanks,
>> Dragomir
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Dragomir Haralambiev

Hi,
When change the answer flag to

$var(rtpengine_flags) = " RTP/SAVP  rtcp-mux-offer ICE=force";
 rtpengine_answer("$var(rtpengine_flags)");

Call is connected but UAC1 not send and receive voices.

Regards,

Dragomir

На ср, 31.07.2019 г. в 15:53 ч. Sasmita Panda  написа:

> Hi Dragomir,
>
> I had mentioned to modify this according to your requirement .   If your
> phone only support RTP/SAVP then change the flag what I have mentioned
> while answering .
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
>
> On Wed, Jul 31, 2019 at 6:17 PM Johan De Clercq  wrote:
>
>> Use rtp/savp
>>
>> On Wed, 31 Jul 2019, 14:40 Dragomir Haralambiev, 
>> wrote:
>>
>>> Hi,
>>>
>>> Thanks for your replay, but this not working.
>>>
>>> UAC1 receive 183 session progress with:
>>> receive audio 50106 UDP/TLS/RTP/SAVP 0 8 18 101
>>>
>>> UAC1   send to Opensips CANCEL.
>>>
>>> I make test with MicroSips latest version.
>>>
>>> Best regards,
>>> Dragomir
>>>
>>> На ср, 31.07.2019 г. в 15:04 ч. Sasmita Panda 
>>> написа:
>>>
 Hi ,

 You have to do something like below  wherever you are calling
 rtpengine_offer/rtpengine_answer.

 $var(rtpengine_flags) = "RTP/AVP replace-session-connection
 replace-origin ICE=remove";
  rtpengine_offer("$var(rtpengine_flags)");

 $var(rtpengine_flags) = "UDP/TLS/RTP/SAVP rtcp-mux-offer ICE=force";
  rtpengine_answer("$var(rtpengine_flags)");

 You can modify this according to your requirement .


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


 On Wed, Jul 31, 2019 at 5:16 PM Dragomir Haralambiev <
 goup2...@gmail.com> wrote:

> Hello,
>
> I have 2 applications connected to Opensips+rtpengine:
> UAC1 -use encryption always. SRTP (RTP/SAVP)
> UAC2 - never use encryption  . RTP (RTP/AVP)
>
> How to setup Opensips to make follow call:
> UAC1 SRTP -> Opensips+rtpengine ---> UAC2 RTP
>
> Thanks,
> Dragomir
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Sasmita Panda

Hi Dragomir,

I had mentioned to modify this according to your requirement .   If your
phone only support RTP/SAVP then change the flag what I have mentioned
while answering .


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


On Wed, Jul 31, 2019 at 6:17 PM Johan De Clercq  wrote:

> Use rtp/savp
>
> On Wed, 31 Jul 2019, 14:40 Dragomir Haralambiev, 
> wrote:
>
>> Hi,
>>
>> Thanks for your replay, but this not working.
>>
>> UAC1 receive 183 session progress with:
>> receive audio 50106 UDP/TLS/RTP/SAVP 0 8 18 101
>>
>> UAC1   send to Opensips CANCEL.
>>
>> I make test with MicroSips latest version.
>>
>> Best regards,
>> Dragomir
>>
>> На ср, 31.07.2019 г. в 15:04 ч. Sasmita Panda 
>> написа:
>>
>>> Hi ,
>>>
>>> You have to do something like below  wherever you are calling
>>> rtpengine_offer/rtpengine_answer.
>>>
>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>>> replace-origin ICE=remove";
>>>  rtpengine_offer("$var(rtpengine_flags)");
>>>
>>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVP rtcp-mux-offer ICE=force";
>>>  rtpengine_answer("$var(rtpengine_flags)");
>>>
>>> You can modify this according to your requirement .
>>>
>>>
>>> *Thanks & Regards*
>>> *Sasmita Panda*
>>> *Senior Network Testing and Software Engineer*
>>> *3CLogic , ph:07827611765*
>>>
>>>
>>> On Wed, Jul 31, 2019 at 5:16 PM Dragomir Haralambiev 
>>> wrote:
>>>
 Hello,

 I have 2 applications connected to Opensips+rtpengine:
 UAC1 -use encryption always. SRTP (RTP/SAVP)
 UAC2 - never use encryption  . RTP (RTP/AVP)

 How to setup Opensips to make follow call:
 UAC1 SRTP -> Opensips+rtpengine ---> UAC2 RTP

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

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

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Johan De Clercq

Use rtp/savp

On Wed, 31 Jul 2019, 14:40 Dragomir Haralambiev,  wrote:

> Hi,
>
> Thanks for your replay, but this not working.
>
> UAC1 receive 183 session progress with:
> receive audio 50106 UDP/TLS/RTP/SAVP 0 8 18 101
>
> UAC1   send to Opensips CANCEL.
>
> I make test with MicroSips latest version.
>
> Best regards,
> Dragomir
>
> На ср, 31.07.2019 г. в 15:04 ч. Sasmita Panda  написа:
>
>> Hi ,
>>
>> You have to do something like below  wherever you are calling
>> rtpengine_offer/rtpengine_answer.
>>
>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>> replace-origin ICE=remove";
>>  rtpengine_offer("$var(rtpengine_flags)");
>>
>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVP rtcp-mux-offer ICE=force";
>>  rtpengine_answer("$var(rtpengine_flags)");
>>
>> You can modify this according to your requirement .
>>
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Senior Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>>
>>
>> On Wed, Jul 31, 2019 at 5:16 PM Dragomir Haralambiev 
>> wrote:
>>
>>> Hello,
>>>
>>> I have 2 applications connected to Opensips+rtpengine:
>>> UAC1 -use encryption always. SRTP (RTP/SAVP)
>>> UAC2 - never use encryption  . RTP (RTP/AVP)
>>>
>>> How to setup Opensips to make follow call:
>>> UAC1 SRTP -> Opensips+rtpengine ---> UAC2 RTP
>>>
>>> Thanks,
>>> Dragomir
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Dragomir Haralambiev

Hi,

Thanks for your replay, but this not working.

UAC1 receive 183 session progress with:
receive audio 50106 UDP/TLS/RTP/SAVP 0 8 18 101

UAC1   send to Opensips CANCEL.

I make test with MicroSips latest version.

Best regards,
Dragomir

На ср, 31.07.2019 г. в 15:04 ч. Sasmita Panda  написа:

> Hi ,
>
> You have to do something like below  wherever you are calling
> rtpengine_offer/rtpengine_answer.
>
> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
> ICE=remove";
>  rtpengine_offer("$var(rtpengine_flags)");
>
> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVP rtcp-mux-offer ICE=force";
>  rtpengine_answer("$var(rtpengine_flags)");
>
> You can modify this according to your requirement .
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
>
> On Wed, Jul 31, 2019 at 5:16 PM Dragomir Haralambiev 
> wrote:
>
>> Hello,
>>
>> I have 2 applications connected to Opensips+rtpengine:
>> UAC1 -use encryption always. SRTP (RTP/SAVP)
>> UAC2 - never use encryption  . RTP (RTP/AVP)
>>
>> How to setup Opensips to make follow call:
>> UAC1 SRTP -> Opensips+rtpengine ---> UAC2 RTP
>>
>> Thanks,
>> Dragomir
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Sasmita Panda

Hi ,

You have to do something like below  wherever you are calling
rtpengine_offer/rtpengine_answer.

$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
 rtpengine_offer("$var(rtpengine_flags)");

$var(rtpengine_flags) = "UDP/TLS/RTP/SAVP rtcp-mux-offer ICE=force";
 rtpengine_answer("$var(rtpengine_flags)");

You can modify this according to your requirement .


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


On Wed, Jul 31, 2019 at 5:16 PM Dragomir Haralambiev 
wrote:

> Hello,
>
> I have 2 applications connected to Opensips+rtpengine:
> UAC1 -use encryption always. SRTP (RTP/SAVP)
> UAC2 - never use encryption  . RTP (RTP/AVP)
>
> How to setup Opensips to make follow call:
> UAC1 SRTP -> Opensips+rtpengine ---> UAC2 RTP
>
> Thanks,
> Dragomir
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] SRTP to RTP

2019-07-31 Thread Dragomir Haralambiev

Hello,

I have 2 applications connected to Opensips+rtpengine:
UAC1 -use encryption always. SRTP (RTP/SAVP)
UAC2 - never use encryption  . RTP (RTP/AVP)

How to setup Opensips to make follow call:
UAC1 SRTP -> Opensips+rtpengine ---> UAC2 RTP

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

[OpenSIPS-Users] CacheDB_Redis authentication

2019-07-31 Thread Yury Kirsanov

Hi,
Could you please let me know correct syntax of modparam for cachedb_redis
for password authentication? Or is it not supported at all? I've tried
different variants but can't get it to work, always getting an error:

Jul 31 19:22:56 ERROR:cachedb_redis:redis_set: Redis operation failure -
0x1b19410 NOAUTH Authentication required.
Jul 31 19:22:56 ERROR:cachedb_redis:redis_set: giving up on query

Thanks!

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

Re: [OpenSIPS-Users] mhomed=1 and force_send_socket()

2019-07-31 Thread Vitalii Aleksandrov

When user is registered everything is OK. The problem appears when I 
send a call to another proxy or carrier. With UDP transport opensips 
correctly detects an outgoing interface and forwards a call. With 
TCP/TLS transport opensips is not that smart. Found the place in code 
which detects an outgoing socket and it just takes the first interface 
for TCP with a comment that it's too complicated to detect it properly.
It means if you have a proxy with multiple LAN/WAN interfaces you can't 
just forward calls to different trunks and have to manually manage 
force_send_socket().
And unfortunately this trick with $fs doesn't work for BIN and I just 
can't have cluster neighbors in different segments of my network.


mhomed =1 will make sure that the outgoing interface towards user A is 
the interface on which user A has registered.

Hence you need to be very very careful with mhomed=1.
I use force_send_socket when I a have a mixed environment (e.g users 
that register, connection to provider without registration, etc).


these are my 2 drops of wisdom :-)

as for BIN, that I can't explain as I never used that type of interface.

.
Op di 30 jul. 2019 om 17:16 schreef Vitalii Aleksandrov 
mailto:vitalik.v...@gmail.com>>:


Hi,

Have a problem with a multihomed proxy which doesn't select outgoing
interface correctly. Found similar topics where people discussed
usage
of force_send_socket() when they want to force opensips to use some
interface.

As far as I understood "mhomed=1" is the option which forces
opensips to
select a proper outgoing interface in multihomed environment. This
options really works, but unfortunately only for UDP. When
opensips has
multiple TCP / TLS / BIN interfaces, opensips just takes the first
from
the list which is not always correct and fails to establish an
outgoing
connection.

Since I have a module that detects an outgoing interface I can use
this
info with $fs to force correct socket for SIP. Unfortunately this
trick
doesn't work for BIN and it's not possible to setup a multihomed
opensips with clustering neighbors in both internal and external
networks.

Is there any workaround for BIN?


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


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

[OpenSIPS-Users] Loglevel priority prefix in /var/log/messages

2019-07-31 Thread Denys Pozniak

Hello!

Please help me to configure loglevel prefix priority (like INFO, NOTICE,
WARNING, ERROR) for /var/log/messages.
As you can see below OpenSIPS does not display it, but for example,
Rtpengine does.

### Global Settings 
log_level=3
log_stderror=no
log_facility=LOG_LOCAL0
...

### Routing Logic 

route{
xlog("L_INFO","info_New message rm=$rm / si=$si / ru=$ru / ci=$ci
\n");
xlog("L_NOTICE","notice_New message rm=$rm / si=$si / ru=$ru /
ci=$ci \n");
xlog("L_WARN","warning_New message rm=$rm / si=$si / ru=$ru /
ci=$ci \n");
xlog("L_ERR","error_New message rm=$rm / si=$si / ru=$ru / ci=$ci
\n");
...

[root@localhost opensips]# tail -f /var/log/messages
Jul 27 15:28:31 localhost /usr/sbin/opensips[32644]: notice_New message
rm=INVITE / si=192.168.56.1 / ru=sip:5@192.168.56.3 / ci=
008BDC86-DDB1-E911-80AA-492E68E5C084@192.168.56.1
Jul 27 15:28:31 localhost /usr/sbin/opensips[32644]: warning_New message
rm=INVITE / si=192.168.56.1 / ru=sip:5@192.168.56.3 / ci=
008BDC86-DDB1-E911-80AA-492E68E5C084@192.168.56.1
Jul 27 15:28:31 localhost /usr/sbin/opensips[32644]: error_New message
rm=INVITE / si=192.168.56.1 / ru=sip:5@192.168.56.3 / ci=
008BDC86-DDB1-E911-80AA-492E68E5C084@192.168.56.1
Jul 27 15:28:31 localhost rtpengine: [1564255711.862367] INFO: [
008BDC86-DDB1-E911-80AA-492E68E5C084@192.168.56.1]: Received command
'offer' from 10.10.200.43:53649
Jul 27 15:28:31 localhost rtpengine: [1564255711.862410] NOTICE: [
008BDC86-DDB1-E911-80AA-492E68E5C084@192.168.56.1]: Creating new call
Jul 27 15:28:31 localhost rtpengine: [1564255711.862684] INFO: [
008BDC86-DDB1-E911-80AA-492E68E5C084@192.168.56.1]: Replying to 'offer'
from 10.10.200.43:53649 (elapsed time 0.000304 sec)

[root@localhost opensips]# opensips -V
version: opensips 3.0.0 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC,
F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
main.c compiled on 13:02:44 May 30 2019 with gcc 4.8.5

[root@localhost opensips]# cat /etc/redhat-release
CentOS Linux release 7.6.1810 (Core)


-- 

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

[OpenSIPS-Users] extra_fields not working in opensips 2.4.x

2019-07-31 Thread J E H A N Z A I B

Hi there,
I used to have old version 1.11.x in the past and extra fields in the acc
module used to work perfectly fine. I have updated my opensips to 2.4 now
and extra_fields are not working.

according to the documentation the from uri and to uri should be put in by
default but its not.

also the extra fields i wanted to write $si value but i cant. i can use
$acc_extra(fu)=$fu; something like this before calling do_accounting but i
do not wanted to put value manually $fu should be put in if we use
extra_fields and add it like i used to do in the 1.11 version.

can anyone help me if i am missing anything.

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

Re: [OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 and 2.4.2

Re: [OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 and 2.4.2

[OpenSIPS-Users] Functional Difference B2B_Bridge between OpenSIP 1.8 and 2.4.2

Re: [OpenSIPS-Users] SRTP to RTP

Re: [OpenSIPS-Users] SRTP to RTP

Re: [OpenSIPS-Users] SRTP to RTP

Re: [OpenSIPS-Users] SRTP to RTP

Re: [OpenSIPS-Users] SRTP to RTP

Re: [OpenSIPS-Users] SRTP to RTP

Re: [OpenSIPS-Users] SRTP to RTP

[OpenSIPS-Users] SRTP to RTP

[OpenSIPS-Users] CacheDB_Redis authentication

Re: [OpenSIPS-Users] mhomed=1 and force_send_socket()

[OpenSIPS-Users] Loglevel priority prefix in /var/log/messages

[OpenSIPS-Users] extra_fields not working in opensips 2.4.x

15 matches

Site Navigation

Mail list logo

Footer information