[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc
eric at uphreak.com
Thu Jun 23 16:28:27 CEST 2016
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";
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
> 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 list
> Users at lists.opensips.org
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