[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[John Nash]

I double checked my rtpengine offer answer calls and now using
https://github.com/onsip/sipjs-examples/tree/master/demo-phone but I face
same issue (no audio either side) and error  "SRTP output wanted, but no
crypto suite was negotiated" Rtpengine also I updated to the latest now.

Am I using correct sip.js example? I copied it to my server and accessing
it using https: (used letsencrypt)

On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme <eric at uphreak.com> wrote:

> 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a
> much more active project that sipml5.
>
> 2. Im guessing that you are not properly passing flags to RTPEngine.  If
> you want to have DTLS-SRTP between the browser, and plain RTP/AVP between
> RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and
> "answer" dtls-srtp back up to the browser.
>
> the offer to freeswitch would be:
>
>         $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
>
>
> and the answer back up to the browswer would be:
>
>         $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
>
>
> -Eric
>
>
>
> On 06/23/2016 08:20 AM, John Nash wrote:
>
> I am following
> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying
> to test a call
>
> sipml5 ----------->Opensips + rtpengine --------> SIP end point
> (Freeswitch)
>
> But I do not have any audio on both sides. I see this error at rtpengine
> log "SRTP output wanted, but no crypto suite was negotiated"
>
> Anyone tested this scenario positive?
>
>
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