[OpenSIPS-Users] SRTP to RTP

[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 <goup2010 at gmail.com>
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.work at 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 <goup2010 at gmail.com>
>> 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 <spanda at 3clogic.com>
>>> 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 <Johan at democon.be>
>>>> wrote:
>>>>
>>>>> Use rtp/savp
>>>>>
>>>>> On Wed, 31 Jul 2019, 14:40 Dragomir Haralambiev, <goup2010 at gmail.com>
>>>>> 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 <spanda at 3clogic.com>
>>>>>> 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 <
>>>>>>> goup2010 at 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
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>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
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-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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