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 <e...@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? > > > _______________________________________________ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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