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

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

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

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)

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

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

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

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,

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

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:

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

[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

[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.

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

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