stream webrtc to rtmp server using ffmpeg

[sameer...@dynaptsolutions.com]

Hi,

I have implemented implemented onetomany tutorial. It's working fine but now i want to stream .sdp file to rtmp server. I don't know the flow how to find sdp file and rtpendpoint. I am quite new to this. Can you please help me out.

[Ivan Gracia]

There is a lot of information on the list about this. Please have a look at the different threads. We can guide you if you are lost somewhere, but it wouldn't make much sense to repeat all the information that is already here in the list ;-)

Ivan Gracia

On Fri, Sep 30, 2016 at 3:22 PM, <sameer...@dynaptsolutions.com> wrote:

Hi,

I have implemented implemented onetomany tutorial. It's working fine but now i want to stream .sdp file to rtmp server. I don't know the flow how to find sdp file and rtpendpoint. I am quite new to this. Can you please help me out.

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[Sameer Saxena]

I have created the sdp using rtp end point but when i process this sdp using ffmpeg i got the following error. I am using the below code for creating sdp file.I get error from ffmpeg:

bind failed: Address already in use

rtp.sdp: Invalid data found when processing input

webRtcEndpoint.processOffer(sdpOffer, function(error, sdpAnswer) {

if (error) {

stop(sessionId);

return callback(error);

}

if (presenter === null) {

stop(sessionId);

return callback(noPresenterMessage);

}

callback(null, sdpAnswer);

});

presenter.pipeline.create('RtpEndpoint',function(error, rtpEndpoint){

    var sdp = 'v=0';

sdp += '\no=- 3641290734 3641290734 IN IP4 127.0.0.1';

sdp += '\ns=media server';

sdp += '\nc=IN IP4 127.0.0.1';

sdp += '\nt=0 0';

sdp += '\nm=audio -audio port- RTP/AVP 0';

sdp += '\na=rtpmap:0 PCMU/8000';

sdp += '\nm=video -video port- RTP/AVP 101';

sdp += '\na=rtpmap:101 H264/90000';

            // rtpEndpoint.setMaxVideoRecvBandwidth(12000);

            // rtpEndpoint.setMaxVideoSendBandwidth(12000);

           rtpEndpoint.processOffer(sdp, function(error, sdpAnswer){

                console.log('Offer:\n', sdp);

                console.log('Answer:\n', sdpAnswer);

            });

            presenter.webRtcEndpoint.connect(rtpEndpoint);

[Ivan Gracia]

You need to add, in the SDP that the RTP endpoint processes, the port where it should be sending media. Probably this post could help you.

Ivan Gracia

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