Problem: notSetRemoteDescription error on phone browsers

[Javier Inh]

Hello, 

As I found in webrtc_adaptor.js code:

/**

                        * This error generally occurs in codec incompatibility.

                        * AMS for a now supports H.264 codec. This error happens when some browsers try to open it from VP8.

*/

So I modified webrtc_adaptor.js to remove VP8 and VP9 video codecs in webrtc sdp:

this.startPublishing = function(idOfStream) {

var streamId = idOfStream;

thiz.initPeerConnection(streamId);

thiz.remotePeerConnection[streamId].createOffer(thiz.sdp_constraints)

.then(function(configuration) {

//CLOUDAWS

configuration.sdp = thiz.removeCodec(configuration.sdp, 'VP8'); //Function to modify the sdp

configuration.sdp = thiz.removeCodec(configuration.sdp, 'VP9'); //Function to modify the sdp

//////

thiz.gotDescription(configuration, streamId);

})

.catch(function (error) {

console.error("create offer error for stream id: " + streamId + " error: " + error);

});

};

//CLOUDAWS

this.removeCodec = function(orgsdp, codec) {

var internalFunc = function(sdp) {

var codecre = new RegExp('(a=rtpmap:(\\d*) ' + codec + '\/90000\\r\\n)');

var rtpmaps = sdp.match(codecre);

if (rtpmaps == null || rtpmaps.length <= 2) {

console.log ("CODEC CLOUDAWS " + sdp);

return sdp;

}

var rtpmap = rtpmaps[2];

// var modsdp = sdp.replace(codecre, "");​

var modsdp = sdp.replace(codecre, "");

var rtcpre = new RegExp('(a=rtcp-fb:' + rtpmap + '.*\r\n)', 'g');

//  modsdp = modsdp.replace(rtcpre, "");​

modsdp = modsdp.replace(rtcpre, "");

var fmtpre = new RegExp('(a=fmtp:' + rtpmap + '.*\r\n)', 'g');

//    modsdp = modsdp.replace(fmtpre, "");​

modsdp = modsdp.replace(fmtpre, "");

var aptpre = new RegExp('(a=fmtp:(\\d*) apt=' + rtpmap + '\\r\\n)');

var aptmaps = modsdp.match(aptpre);

var fmtpmap = "";

if (aptmaps != null && aptmaps.length >= 3) {

fmtpmap = aptmaps[2];

// modsdp = modsdp.replace(aptpre, "");​

modsdp = modsdp.replace(aptpre, "");

var rtppre = new RegExp('(a=rtpmap:' + fmtpmap + '.*\r\n)', 'g');

modsdp = modsdp.replace(rtppre, "");

}

var videore = /(m=video.*\r\n)/;

var videolines = modsdp.match(videore);

if (videolines != null) {

//If many m=video are found in SDP, this program doesn't work.

var videoline = videolines[0].substring(0, videolines[0].length - 2);

var videoelem = videoline.split(" ");

var modvideoline = videoelem[0];

for (var i = 1; i < videoelem.length; i++) {

if (videoelem[i] == rtpmap || videoelem[i] == fmtpmap) {

continue;

}

modvideoline += " " + videoelem[i];

}

modvideoline += "\r\n";

modsdp = modsdp.replace(videore, modvideoline);

}

return internalFunc(modsdp);

};

return internalFunc(orgsdp);

}

With these code I achieved to remove codecs. Resulting offer:

v=0

o=- 1336756143568039701 2 IN IP4 127.0.0.1

s=-

t=0 0

a=group:BUNDLE 0 1

a=msid-semantic: WMS gEowYlm5gIr0TcLmrHg9tD50DYRWSRYCRvun

m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126

c=IN IP4 0.0.0.0

a=rtcp:9 IN IP4 0.0.0.0

a=ice-ufrag:oQ6v

a=ice-pwd:eSGvCPCWa02NccxozCLAQfcN

a=ice-options:trickle

a=fingerprint:sha-256 92:F6:E1:EE:65:F5:83:0B:CA:4C:6E:FA:1D:F9:93:38:7B:8E:21:69:6F:08:06:26:C8:EE:F4:0D:E7:3D:F3:A4

a=setup:actpass

a=mid:0

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level

a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01

a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid

a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id

a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id

a=sendrecv

a=msid:gEowYlm5gIr0TcLmrHg9tD50DYRWSRYCRvun 88f3d4d2-4ee7-42a1-a4e7-a49425dfa5aa

a=rtcp-mux

a=rtpmap:111 opus/48000/2

a=rtcp-fb:111 transport-cc

a=fmtp:111 minptime=10;useinbandfec=1

a=rtpmap:103 ISAC/16000

a=rtpmap:104 ISAC/32000

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:106 CN/32000

a=rtpmap:105 CN/16000

a=rtpmap:13 CN/8000

a=rtpmap:110 telephone-event/48000

a=rtpmap:112 telephone-event/32000

a=rtpmap:113 telephone-event/16000

a=rtpmap:126 telephone-event/8000

a=ssrc:3330065160 cname:qBHnNONNLZ0u/GEQ

a=ssrc:3330065160 msid:gEowYlm5gIr0TcLmrHg9tD50DYRWSRYCRvun 88f3d4d2-4ee7-42a1-a4e7-a49425dfa5aa

a=ssrc:3330065160 mslabel:gEowYlm5gIr0TcLmrHg9tD50DYRWSRYCRvun

a=ssrc:3330065160 label:88f3d4d2-4ee7-42a1-a4e7-a49425dfa5aa

m=video 9 UDP/TLS/RTP/SAVPF 102 122 127 121 125 107 108 109 124 120 123 119 114 115 116

c=IN IP4 0.0.0.0

a=rtcp:9 IN IP4 0.0.0.0

a=ice-ufrag:oQ6v

a=ice-pwd:eSGvCPCWa02NccxozCLAQfcN

a=ice-options:trickle

a=fingerprint:sha-256 92:F6:E1:EE:65:F5:83:0B:CA:4C:6E:FA:1D:F9:93:38:7B:8E:21:69:6F:08:06:26:C8:EE:F4:0D:E7:3D:F3:A4

a=setup:actpass

a=mid:1

a=extmap:14 urn:ietf:params:rtp-hdrext:toffset

a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

a=extmap:13 urn:3gpp:video-orientation

a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01

a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay

a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type

a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing

a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07

a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space

a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid

a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id

a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id

a=sendrecv

a=msid:gEowYlm5gIr0TcLmrHg9tD50DYRWSRYCRvun 1875d522-7c14-4044-9540-244f2ac19f71

a=rtcp-mux

a=rtcp-rsize

a=rtpmap:102 H264/90000

a=rtcp-fb:102 goog-remb

a=rtcp-fb:102 transport-cc

a=rtcp-fb:102 ccm fir

a=rtcp-fb:102 nack

a=rtcp-fb:102 nack pli

a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f

a=rtpmap:122 rtx/90000

a=fmtp:122 apt=102

a=rtpmap:127 H264/90000

a=rtcp-fb:127 goog-remb

a=rtcp-fb:127 transport-cc

a=rtcp-fb:127 ccm fir

a=rtcp-fb:127 nack

a=rtcp-fb:127 nack pli

a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f

a=rtpmap:121 rtx/90000

a=fmtp:121 apt=127

a=rtpmap:125 H264/90000

a=rtcp-fb:125 goog-remb

a=rtcp-fb:125 transport-cc

a=rtcp-fb:125 ccm fir

a=rtcp-fb:125 nack

a=rtcp-fb:125 nack pli

a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f

a=rtpmap:107 rtx/90000

a=fmtp:107 apt=125

a=rtpmap:108 H264/90000

a=rtcp-fb:108 goog-remb

a=rtcp-fb:108 transport-cc

a=rtcp-fb:108 ccm fir

a=rtcp-fb:108 nack

a=rtcp-fb:108 nack pli

a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f

a=rtpmap:109 rtx/90000

a=fmtp:109 apt=108

a=rtpmap:124 H264/90000

a=rtcp-fb:124 goog-remb

a=rtcp-fb:124 transport-cc

a=rtcp-fb:124 ccm fir

a=rtcp-fb:124 nack

a=rtcp-fb:124 nack pli

a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032

a=rtpmap:120 rtx/90000

a=fmtp:120 apt=124

a=rtpmap:123 H264/90000

a=rtcp-fb:123 goog-remb

a=rtcp-fb:123 transport-cc

a=rtcp-fb:123 ccm fir

a=rtcp-fb:123 nack

a=rtcp-fb:123 nack pli

a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032

a=rtpmap:119 rtx/90000

a=fmtp:119 apt=123

a=rtpmap:114 red/90000

a=rtpmap:115 rtx/90000

a=fmtp:115 apt=114

a=rtpmap:116 ulpfec/90000

a=ssrc-group:FID 1760319672 3497816362

a=ssrc:1760319672 cname:qBHnNONNLZ0u/GEQ

a=ssrc:1760319672 msid:gEowYlm5gIr0TcLmrHg9tD50DYRWSRYCRvun 1875d522-7c14-4044-9540-244f2ac19f71

a=ssrc:1760319672 mslabel:gEowYlm5gIr0TcLmrHg9tD50DYRWSRYCRvun

a=ssrc:1760319672 label:1875d522-7c14-4044-9540-244f2ac19f71

a=ssrc:3497816362 cname:qBHnNONNLZ0u/GEQ

a=ssrc:3497816362 msid:gEowYlm5gIr0TcLmrHg9tD50DYRWSRYCRvun 1875d522-7c14-4044-9540-244f2ac19f71

a=ssrc:3497816362 mslabel:gEowYlm5gIr0TcLmrHg9tD50DYRWSRYCRvun

a=ssrc:3497816362 label:1875d522-7c14-4044-9540-244f2ac19f71

BUT:

I´m still having notSetRemoteDescription error on phone browsers

Any idea how to resolve this problem?

I´m not developer so I don't have enough knowlement about.

[mahmut asım Onat]

Hi,

Can you check whether your browser supports H.264 from this link: "https://mozilla.github.io/webrtc-landing/pc_test_no_h264.html"

Best,
Asim

[Javier Inh]

Hi,

Test failed, tested in two phones...

So I don´t understand why Webrtc demo is working (I can stream) in the two same smartphones H.264 is not present: https://webrtc.github.io/samples/src/content/peerconnection/pc1/

Any idea who can I achieve Webrtc streaming from phones (Android 9/10) with Ant Media?

I tested other Webrtc solutions and them works ok.

Thanks.

[Marco Schiattarella]

did you find a solutions? i'm having the same problem!

[Javier Inh]

No, the only option for me has been to use other streaming solution.

[Marco Schiattarella]

i'm doing the same, thank you