subject:"\[OpenSIPS\-Users\] SRTP to RTP"

Re: [OpenSIPS-Users] SRTP to RTP

2019-08-01 Thread David Villasmil

Can you please share a pcap?

On Thu, 1 Aug 2019 at 13:40, Dragomir Haralambiev 
wrote:

> Hi,
>
> I check this. All like OK. Here is SIP flow
>
> 1. tpengine_offer
>
> UAC1 SRTP ---INVITE >  Opensips+rtpengine
> *audio 4004 RTP/SAVP 8 0 18 101*
> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
> ICE=remove";
> tpengine_offer("$var(rtpengine_flags)");
> Opensips+rtpengine - INVITE --> UAC2 RTP
> *audio 50190 RTP/AVP 8 0 18 101 *
>
> 2. rtpengine_answer when receive 183 (Early Media)
> *audio 15612 RTP/AVP 0 8 18 101*
> Opensips+rtpengine < 183 Early Media  UAC2 RTP
> $var(rtpengine_flags) = "RTP/SAVP replace-session-connection
> replace-origin ICE=force";
> rtpengine_answer("$var(rtpengine_flags)");
>
> 3. rtpengine_answer when receive 200 OK
> *audio 15612 RTP/AVP 0 8 18 101*
> Opensips+rtpengine < 200 OK UAC2 RTP
> $var(rtpengine_flags) = "RTP/SAVP replace-session-connection
> replace-origin ICE=force";
> rtpengine_answer("$var(rtpengine_flags)");
> UAC1 <- 200 OK - Opensips+rtpengine
> *audio 50208 RTP/SAVP 0 8 18 101*
>
>
> 4. tpengine_offer when receive re-INVITE
> UAC1 SRTP ---INVITE >  Opensips+rtpengine
> *audio 4004 RTP/SAVP 0 101*
> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
> ICE=remove";
> tpengine_offer("$var(rtpengine_flags)");
> Opensips+rtpengine - INVITE --> UAC2 RTP
> *audio 50190 RTP/AVP 0 101*
>
> На чт, 1.08.2019 г. в 15:16 ч. David Villasmil <
> david.villasmil.w...@gmail.com> написа:
>
>> You must check your SDPs, verify all going to srtp is indeed SRTP SDP.
>> And all going to UAC is not SRTP
>>
>> On Thu, 1 Aug 2019 at 11:59, Dragomir Haralambiev 
>> wrote:
>>
>>> Hi,
>>>
>>> 1. tpengine_offer
>>>
>>> UAC1 SRTP ---INVITE >  Opensips+rtpengine
>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>>> replace-origin ICE=remove";
>>> tpengine_offer("$var(rtpengine_flags)");
>>> Opensips+rtpengine - INVITE --> UAC2 RTP
>>>
>>>
>>> 2. rtpengine_answer when receive 183 (Early Media)
>>> Opensips+rtpengine < 183 Early Media  UAC2 RTP
>>> $var(rtpengine_flags) = "RTP/SAVP replace-session-connection
>>> replace-origin ICE=force";
>>> rtpengine_answer("$var(rtpengine_flags)");
>>>
>>> 3. rtpengine_answer when receive 200 OK
>>> Opensips+rtpengine < 200 OK UAC2 RTP
>>> $var(rtpengine_flags) = "RTP/SAVP replace-session-connection
>>> replace-origin ICE=force";
>>> rtpengine_answer("$var(rtpengine_flags)");
>>>
>>> 4. tpengine_offer when receive re-INVITE
>>> UAC1 SRTP ---INVITE >  Opensips+rtpengine
>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>>> replace-origin ICE=remove";
>>> tpengine_offer("$var(rtpengine_flags)");
>>> Opensips+rtpengine - INVITE --> UAC2 RTP
>>>
>>>
>>> In this case UAC1 SRTP not receive voice.
>>>
>>> Best regards,
>>> Dragomir
>>>
>>> On Wed, Jul 31, 2019, 16:15 David Villasmil <
>>> david.villasmil.w...@gmail.com> 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";
>

Re: [OpenSIPS-Users] SRTP to RTP

2019-08-01 Thread Dragomir Haralambiev

Hi,

I check this. All like OK. Here is SIP flow

1. tpengine_offer

UAC1 SRTP ---INVITE >  Opensips+rtpengine
*audio 4004 RTP/SAVP 8 0 18 101*
$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
tpengine_offer("$var(rtpengine_flags)");
Opensips+rtpengine - INVITE --> UAC2 RTP
*audio 50190 RTP/AVP 8 0 18 101 *

2. rtpengine_answer when receive 183 (Early Media)
*audio 15612 RTP/AVP 0 8 18 101*
Opensips+rtpengine < 183 Early Media  UAC2 RTP
$var(rtpengine_flags) = "RTP/SAVP replace-session-connection replace-origin
ICE=force";
rtpengine_answer("$var(rtpengine_flags)");

3. rtpengine_answer when receive 200 OK
*audio 15612 RTP/AVP 0 8 18 101*
Opensips+rtpengine < 200 OK UAC2 RTP
$var(rtpengine_flags) = "RTP/SAVP replace-session-connection replace-origin
ICE=force";
rtpengine_answer("$var(rtpengine_flags)");
UAC1 <- 200 OK - Opensips+rtpengine
*audio 50208 RTP/SAVP 0 8 18 101*


4. tpengine_offer when receive re-INVITE
UAC1 SRTP ---INVITE >  Opensips+rtpengine
*audio 4004 RTP/SAVP 0 101*
$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
tpengine_offer("$var(rtpengine_flags)");
Opensips+rtpengine - INVITE --> UAC2 RTP
*audio 50190 RTP/AVP 0 101*

На чт, 1.08.2019 г. в 15:16 ч. David Villasmil <
david.villasmil.w...@gmail.com> написа:

> You must check your SDPs, verify all going to srtp is indeed SRTP SDP. And
> all going to UAC is not SRTP
>
> On Thu, 1 Aug 2019 at 11:59, Dragomir Haralambiev 
> wrote:
>
>> Hi,
>>
>> 1. tpengine_offer
>>
>> UAC1 SRTP ---INVITE >  Opensips+rtpengine
>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>> replace-origin ICE=remove";
>> tpengine_offer("$var(rtpengine_flags)");
>> Opensips+rtpengine - INVITE --> UAC2 RTP
>>
>>
>> 2. rtpengine_answer when receive 183 (Early Media)
>> Opensips+rtpengine < 183 Early Media  UAC2 RTP
>> $var(rtpengine_flags) = "RTP/SAVP replace-session-connection
>> replace-origin ICE=force";
>> rtpengine_answer("$var(rtpengine_flags)");
>>
>> 3. rtpengine_answer when receive 200 OK
>> Opensips+rtpengine < 200 OK UAC2 RTP
>> $var(rtpengine_flags) = "RTP/SAVP replace-session-connection
>> replace-origin ICE=force";
>> rtpengine_answer("$var(rtpengine_flags)");
>>
>> 4. tpengine_offer when receive re-INVITE
>> UAC1 SRTP ---INVITE >  Opensips+rtpengine
>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>> replace-origin ICE=remove";
>> tpengine_offer("$var(rtpengine_flags)");
>> Opensips+rtpengine - INVITE --> UAC2 RTP
>>
>>
>> In this case UAC1 SRTP not receive voice.
>>
>> Best regards,
>> Dragomir
>>
>> On Wed, Jul 31, 2019, 16:15 David Villasmil <
>> david.villasmil.w...@gmail.com> 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 &

Re: [OpenSIPS-Users] SRTP to RTP

2019-08-01 Thread David Villasmil

You must check your SDPs, verify all going to srtp is indeed SRTP SDP. And
all going to UAC is not SRTP

On Thu, 1 Aug 2019 at 11:59, Dragomir Haralambiev 
wrote:

> Hi,
>
> 1. tpengine_offer
>
> UAC1 SRTP ---INVITE >  Opensips+rtpengine
> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
> ICE=remove";
> tpengine_offer("$var(rtpengine_flags)");
> Opensips+rtpengine - INVITE --> UAC2 RTP
>
>
> 2. rtpengine_answer when receive 183 (Early Media)
> Opensips+rtpengine < 183 Early Media  UAC2 RTP
> $var(rtpengine_flags) = "RTP/SAVP replace-session-connection
> replace-origin ICE=force";
> rtpengine_answer("$var(rtpengine_flags)");
>
> 3. rtpengine_answer when receive 200 OK
> Opensips+rtpengine < 200 OK UAC2 RTP
> $var(rtpengine_flags) = "RTP/SAVP replace-session-connection
> replace-origin ICE=force";
> rtpengine_answer("$var(rtpengine_flags)");
>
> 4. tpengine_offer when receive re-INVITE
> UAC1 SRTP ---INVITE >  Opensips+rtpengine
> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
> ICE=remove";
> tpengine_offer("$var(rtpengine_flags)");
> Opensips+rtpengine - INVITE --> UAC2 RTP
>
>
> In this case UAC1 SRTP not receive voice.
>
> Best regards,
> Dragomir
>
> On Wed, Jul 31, 2019, 16:15 David Villasmil <
> david.villasmil.w...@gmail.com> 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

Re: [OpenSIPS-Users] SRTP to RTP

2019-08-01 Thread Dragomir Haralambiev

Hi,

1. tpengine_offer

UAC1 SRTP ---INVITE >  Opensips+rtpengine
$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
tpengine_offer("$var(rtpengine_flags)");
Opensips+rtpengine - INVITE --> UAC2 RTP


2. rtpengine_answer when receive 183 (Early Media)
Opensips+rtpengine < 183 Early Media  UAC2 RTP
$var(rtpengine_flags) = "RTP/SAVP replace-session-connection replace-origin
ICE=force";
rtpengine_answer("$var(rtpengine_flags)");

3. rtpengine_answer when receive 200 OK
Opensips+rtpengine < 200 OK UAC2 RTP
$var(rtpengine_flags) = "RTP/SAVP replace-session-connection replace-origin
ICE=force";
rtpengine_answer("$var(rtpengine_flags)");

4. tpengine_offer when receive re-INVITE
UAC1 SRTP ---INVITE >  Opensips+rtpengine
$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
tpengine_offer("$var(rtpengine_flags)");
Opensips+rtpengine - INVITE --> UAC2 RTP


In this case UAC1 SRTP not receive voice.

Best regards,
Dragomir

On Wed, Jul 31, 2019, 16:15 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
>

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
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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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
___
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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
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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
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[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
___
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[OpenSIPS-Users] SRTP to RTP with HardWare

2016-01-18 Thread Dave Lechasseur

Hi,

I want to be able to convert SRTP <-> RTP with my OpenSIPS server.
I understand that OpenSIPS don’t handle (S)RTP.

Which application should I use for this goal and is it possible to use a 
hardware equipment to do it (if yes, do you have any recommendation).

Thank you,

Dave L.
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Re: [OpenSIPS-Users] SRTP to RTP with HardWare

2016-01-18 Thread Dave Lechasseur

Thank you, I'll take a look at it for sure.
If I want to use hardware for the decryption what should I use?

Dave L.

On Mon, Jan 18, 2016 at 12:36 PM -0800, "Tito Cumpen" 
> wrote:

Dave,

Look into rtpengine I believe it can facilitate your requirement.

https://github.com/sipwise/rtpengine
http://www.opensips.org/html/docs/modules/2.1.x/rtpengine

On Mon, Jan 18, 2016 at 1:11 PM, Dave Lechasseur 
> wrote:
Hi,

I want to be able to convert SRTP <-> RTP with my OpenSIPS server.
I understand that OpenSIPS don't handle (S)RTP.

Which application should I use for this goal and is it possible to use a 
hardware equipment to do it (if yes, do you have any recommendation).

Thank you,

Dave L.

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Re: [OpenSIPS-Users] SRTP to RTP with HardWare

2016-01-18 Thread Tito Cumpen

Dave,


Look into rtpengine I believe it can facilitate your requirement.

https://github.com/sipwise/rtpengine
http://www.opensips.org/html/docs/modules/2.1.x/rtpengine

On Mon, Jan 18, 2016 at 1:11 PM, Dave Lechasseur <
dave.lechass...@sbktelecom.com> wrote:

> Hi,
>
> I want to be able to convert SRTP <-> RTP with my OpenSIPS server.
> I understand that OpenSIPS don’t handle (S)RTP.
>
> Which application should I use for this goal and is it possible to use a
> hardware equipment to do it (if yes, do you have any recommendation).
>
> Thank you,
>
> Dave L.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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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

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] SRTP to RTP with HardWare

Re: [OpenSIPS-Users] SRTP to RTP with HardWare

Re: [OpenSIPS-Users] SRTP to RTP with HardWare

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