[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Eric Tamme]

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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