[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

Eric Tamme 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 
> (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 list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20160623/4f8643ab/attachment.htm>

More information about the Users mailing list