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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