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
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
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
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)
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
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
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
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,
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
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
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
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.
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
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
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