gstreamer with offer sent by browser not working

94 Views Asked by At

I'm trying to modify some code that previously sent an offer from a python client to the browser, now I'd like to switch that relationship and have the browser send the offer. I feel like I'm failing at the last hurdle. The browser is successfully creating an offer:

v=0
o=- 5170002668192921444 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=extmap-allow-mixed
a=msid-semantic: WMS
m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:1r66
a=ice-pwd:z/Lcze20aL4JGekMH8H+YDjq
a=ice-options:trickle
a=fingerprint:sha-256 34:86:C6:69:30:32:C4:42:7E:3A:1F:D8:52:07:3B:AF:D3:45:BE:5D:19:EB:34:DE:1A:7B:40:8C:87:5D:8D:18
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=recvonly
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 35 36 37 38 102 103 104 105 106 107 108 109 127 125 39 40 41 42 43 44 45 46 47 48 112 113 114 115 116 117 118 49
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:1r66
a=ice-pwd:z/Lcze20aL4JGekMH8H+YDjq
a=ice-options:trickle
a=fingerprint:sha-256 34:86:C6:69:30:32:C4:42:7E:3A:1F:D8:52:07:3B:AF:D3:45:BE:5D:19:EB:34:DE:1A:7B:40:8C:87:5D:8D:18
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:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:6 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://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=recvonly
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 VP9/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 profile-id=2
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:35 VP9/90000
a=rtcp-fb:35 goog-remb
a=rtcp-fb:35 transport-cc
a=rtcp-fb:35 ccm fir
a=rtcp-fb:35 nack
a=rtcp-fb:35 nack pli
a=fmtp:35 profile-id=1
a=rtpmap:36 rtx/90000
a=fmtp:36 apt=35
a=rtpmap:37 VP9/90000
a=rtcp-fb:37 goog-remb
a=rtcp-fb:37 transport-cc
a=rtcp-fb:37 ccm fir
a=rtcp-fb:37 nack
a=rtcp-fb:37 nack pli
a=fmtp:37 profile-id=3
a=rtpmap:38 rtx/90000
a=fmtp:38 apt=37
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:103 rtx/90000
a=fmtp:103 apt=102
a=rtpmap:104 H264/90000
a=rtcp-fb:104 goog-remb
a=rtcp-fb:104 transport-cc
a=rtcp-fb:104 ccm fir
a=rtcp-fb:104 nack
a=rtcp-fb:104 nack pli
a=fmtp:104 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:105 rtx/90000
a=fmtp:105 apt=104
a=rtpmap:106 H264/90000
a=rtcp-fb:106 goog-remb
a=rtcp-fb:106 transport-cc
a=rtcp-fb:106 ccm fir
a=rtcp-fb:106 nack
a=rtcp-fb:106 nack pli
a=fmtp:106 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=106
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: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=1;profile-level-id=4d001f
a=rtpmap:125 rtx/90000
a=fmtp:125 apt=127
a=rtpmap:39 H264/90000
a=rtcp-fb:39 goog-remb
a=rtcp-fb:39 transport-cc
a=rtcp-fb:39 ccm fir
a=rtcp-fb:39 nack
a=rtcp-fb:39 nack pli
a=fmtp:39 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=4d001f
a=rtpmap:40 rtx/90000
a=fmtp:40 apt=39
a=rtpmap:41 H264/90000
a=rtcp-fb:41 goog-remb
a=rtcp-fb:41 transport-cc
a=rtcp-fb:41 ccm fir
a=rtcp-fb:41 nack
a=rtcp-fb:41 nack pli
a=fmtp:41 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=f4001f
a=rtpmap:42 rtx/90000
a=fmtp:42 apt=41
a=rtpmap:43 H264/90000
a=rtcp-fb:43 goog-remb
a=rtcp-fb:43 transport-cc
a=rtcp-fb:43 ccm fir
a=rtcp-fb:43 nack
a=rtcp-fb:43 nack pli
a=fmtp:43 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=f4001f
a=rtpmap:44 rtx/90000
a=fmtp:44 apt=43
a=rtpmap:45 AV1/90000
a=rtcp-fb:45 goog-remb
a=rtcp-fb:45 transport-cc
a=rtcp-fb:45 ccm fir
a=rtcp-fb:45 nack
a=rtcp-fb:45 nack pli
a=rtpmap:46 rtx/90000
a=fmtp:46 apt=45
a=rtpmap:47 AV1/90000
a=rtcp-fb:47 goog-remb
a=rtcp-fb:47 transport-cc
a=rtcp-fb:47 ccm fir
a=rtcp-fb:47 nack
a=rtcp-fb:47 nack pli
a=fmtp:47 profile=1
a=rtpmap:48 rtx/90000
a=fmtp:48 apt=47
a=rtpmap:112 H264/90000
a=rtcp-fb:112 goog-remb
a=rtcp-fb:112 transport-cc
a=rtcp-fb:112 ccm fir
a=rtcp-fb:112 nack
a=rtcp-fb:112 nack pli
a=fmtp:112 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f
a=rtpmap:113 rtx/90000
a=fmtp:113 apt=112
a=rtpmap:114 H264/90000
a=rtcp-fb:114 goog-remb
a=rtcp-fb:114 transport-cc
a=rtcp-fb:114 ccm fir
a=rtcp-fb:114 nack
a=rtcp-fb:114 nack pli
a=fmtp:114 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=64001f
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 red/90000
a=rtpmap:117 rtx/90000
a=fmtp:117 apt=116
a=rtpmap:118 ulpfec/90000
a=rtpmap:49 flexfec-03/90000
a=rtcp-fb:49 goog-remb
a=rtcp-fb:49 transport-cc
a=fmtp:49 repair-window=10000000

and the python code looks like it is successfully creating an answer:

v=0
o=- 5170002668192921444 2 IN IP4 0.0.0.0
s=-
t=0 0
a=group:BUNDLE 0 1
m=audio 9 UDP/TLS/RTP/SAVPF 111
c=IN IP4 0.0.0.0
a=ice-ufrag:KP2HFWHaxD6+Z5yp29TnPYbyxTHMUrrj
a=ice-pwd:CsPRE7zKtZGEuWGfHWbyPBANQ0MJsY2d
a=mid:0
a=rtcp-mux
a=setup:active
a=rtpmap:111 OPUS/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=inactive
a=fingerprint:sha-256 52:36:FF:99:CE:00:7D:54:81:D9:BB:F4:48:59:15:38:D6:35:00:C4:C3:07:7C:F1:FF:37:5B:BD:47:5E:69:B9
m=video 9 UDP/TLS/RTP/SAVPF 102
c=IN IP4 0.0.0.0
a=ice-ufrag:KP2HFWHaxD6+Z5yp29TnPYbyxTHMUrrj
a=ice-pwd:CsPRE7zKtZGEuWGfHWbyPBANQ0MJsY2d
a=mid:1
a=rtcp-mux
a=setup:active
a=rtpmap:102 H264/90000
a=rtcp-fb:102 nack pli
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 transport-cc
a=fmtp:102 packetization-mode=1;level-asymmetry-allowed=1
a=sendonly
a=fingerprint:sha-256 52:36:FF:99:CE:00:7D:54:81:D9:BB:F4:48:59:15:38:D6:35:00:C4:C3:07:7C:F1:FF:37:5B:BD:47:5E:69:B9

In the browser I even see the track getting added to the video control, but then the video control just displays a spinning disc and never displays the video. After a second or two the browser will report connectionstatechange failed.

My pipeline is created with the following string:

webrtcbin bundle-policy=max-bundle name=sink stun-server=stun://stun.kinesisvideo.us-west-2.amazonaws.com:443
appsrc name=source ! videoconvert ! videoscale ! video/x-raw,width=1280,height=720 ! videorate ! video/x-raw,framerate=25/1 ! textoverlay name=overlay font-desc=\"Sans 24\" ! queue ! x264enc speed-preset=ultrafast tune=zerolatency bframes=0 key-int-max=25 ! video/x-h264,profile=baseline ! rtph264pay !
queue ! application/x-rtp,media=video,encoding-name=H264 ! sink.

My pipeline is setup like this:

def setup_gstreamer(self):
    try:
        # Create a new pipeline

        self.pipeline = Gst.parse_launch(self.pipeline_description)
        self.appsrc = self.pipeline.get_by_name("source")
        self.webrtc = self.pipeline.get_by_name("sink")
        self.overlay = self.pipeline.get_by_name("overlay")

        # Set source caps
        appsrc_caps = Gst.Caps.from_string(
            f"video/x-raw,format=RGB,width={self.source_width},height={self.source_height},framerate={self.source_framerate}/1"
        )
        self.appsrc.set_property("caps", appsrc_caps)

        # Set source properties
        self.appsrc.set_property("format", Gst.Format.TIME)
        self.appsrc.set_property("is-live", True)

        # Attach signals
        self.webrtc.connect("on-negotiation-needed", self.on_negotiation_needed)
        self.webrtc.connect("on-ice-candidate", self.send_ice_candidate_message)
        self.webrtc.connect("notify::ice-gathering-state", self.on_ice_gathering_state_notify)

        self.pipeline.set_state(Gst.State.READY)

        self.webrtc.connect("pad-added", self.on_incoming_stream, self.pipeline)

        self.get_ice_candidates()

        print("Starting pipeline")

    except Exception as ex:
        raise Exception("Failed to start pipeline", ex)

    ret = self.pipeline.set_state(Gst.State.PLAYING)
    if ret == Gst.StateChangeReturn.FAILURE:
        raise Exception("Failed to start pipeline")

When the offer is received I create an answer:

def set_offer_on_gstreamer(self, sdp):
    res, sdpmsg = GstSdp.SDPMessage.new()
    GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
    offer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
    promise = Gst.Promise.new()
    self.webrtc.emit("set-remote-description", offer, promise)
    promise.wait()

    promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None)
    self.webrtc.emit("create-answer", None, promise)

def on_answer_created(self, promise, _, __):
    reply = promise.get_reply()
    answer = reply.get_value("answer")
    self.webrtc.emit("set-local-description", answer, None)

    answer_text = answer.sdp.as_text()
    # answer is then sent to the signaling channel

How can I debug this further? What might be going on?

0

There are 0 best solutions below