AudioBridge room with allow_rtp_participants enable not working

General
1

I’ve run into a problem with integrating janus with a sip server. I have been following this guide. I have a asterisk server as sip server. I see all the connections have been setup. When I make a sip call, this function invoked

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then I using janode to join a audiobridge room with allow_rtp_participants enable. The join payload have similar format to this one

{
        "request" : "join",
        [..]
        "rtp" : {
                "ip" : "<IP address you want media to be sent to>",
                "port" : <port you want media to be sent to>,
                "payload_type" : <payload type to use for RTP packets (optional; only needed in case Opus is used, automatic for G.711)>,
                "audiolevel_ext" : <ID of the audiolevel RTP extension, if used (optional)>,
                "fec" : <true|false, whether FEC should be enabled for the Opus stream (optional; only needed in case Opus is used)>
        }
}

. But the response from janus don’t have the rtp property. I wonder if janus need to do any kind of verify the port and ip from request join payload to send back the client the rtp ip and port of server.

2

You need to enable RTP participants in rooms, it’s disabled by default:

3

Thanks you for your response. I have setup a room with allow_rtp_participants set to true, but the response from join request still lack of rtp property. My join detail is something like this
{ room: 1234, display: from, rtp_participant: { ip: ip, port: port, audiolevel_ext: 1, payload_type: 111, fec: true } }
Do you have any idea for this situation?

4

There’s no such thing as rtp_participant, the object must be called rtp.