Failed to set remote video description send parameters

[Igrium]

I'm trying to get node-webrtc to communicate with Unreal Engine, but when I call setRemoteDescription, It's crashing with:

[DOMException [InvalidAccessError]: Failed to set remote offer sdp: Failed to set remote video description send parameters for m-section with mid='0'.]

The SDP it's received is as follows:

v=0
o=- 2972928273947422983 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1 2
a=extmap-allow-mixed
a=msid-semantic: WMS pixelstreaming_audio_stream_id pixelstreaming_video_stream_id
m=video 9 UDP/TLS/RTP/SAVPF 127 97 98 99 100 101 102
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:bNw+
a=ice-pwd:Ig0eXmDSregiA/vvnhMgoU4s
a=ice-options:trickle
a=fingerprint:sha-256 45:16:B6:6D:F2:3E:FB:D0:09:41:03:91:24:74:DD:D8:4D:55:88:5B:46:83:35:82:B9:99:3E:1F:CA:7A:58:84
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 urn:3gpp:video-orientation
a=extmap:4 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:9 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=sendrecv
a=msid:pixelstreaming_video_stream_id video
a=rtcp-mux
a=rtcp-rsize
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=42e01f
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=127
a=rtpmap:98 H264/90000
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 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 red/90000
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 ulpfec/90000
a=ssrc-group:FID 2079964708 2769205221
a=ssrc:2079964708 cname:R5IjzVnJa4ND6xzi
a=ssrc:2079964708 msid:pixelstreaming_video_stream_id video
a=ssrc:2769205221 cname:R5IjzVnJa4ND6xzi
a=ssrc:2769205221 msid:pixelstreaming_video_stream_id video
m=audio 9 UDP/TLS/RTP/SAVPF 96 110
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:bNw+
a=ice-pwd:Ig0eXmDSregiA/vvnhMgoU4s
a=ice-options:trickle
a=fingerprint:sha-256 45:16:B6:6D:F2:3E:FB:D0:09:41:03:91:24:74:DD:D8:4D:55:88:5B:46:83:35:82:B9:99:3E:1F:CA:7A:58:84
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:pixelstreaming_audio_stream_id audio
a=rtcp-mux
a=rtpmap:96 OPUS/48000/2
a=rtcp-fb:96 transport-cc
a=rtpmap:110 telephone-event/48000
a=ssrc:1708106480 cname:R5IjzVnJa4ND6xzi
a=ssrc:1708106480 msid:pixelstreaming_audio_stream_id audio
m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=ice-ufrag:bNw+
a=ice-pwd:Ig0eXmDSregiA/vvnhMgoU4s
a=ice-options:trickle
a=fingerprint:sha-256 45:16:B6:6D:F2:3E:FB:D0:09:41:03:91:24:74:DD:D8:4D:55:88:5B:46:83:35:82:B9:99:3E:1F:CA:7A:58:84
a=setup:actpass
a=mid:2
a=sctp-port:5000
a=max-message-size:262144

I'm not really sure what to do about this, as it seems like nothing I do will make it accept this SDP. The current function to handle it looks like this:

    private async onOffer(message: Messages.offer) {
        if (this.peer && this.peer.connectionState != 'disconnected') {
            console.warn("Streamer is already connected via RTC.");
            this.peer.close();
            this.peer = undefined;
        }
        
        const peer = new wrtc.RTCPeerConnection({
            iceServers: [{ urls: "stun:stunprotocol.org" }]
        });

        const audioTransceiver = peer.addTransceiver('audio');
        const videoTransceiver = peer.addTransceiver('video', { direction: 'sendrecv' });
        peer.createDataChannel("datachannel");

        console.log("Received offer SDP: ", message.sdp);

        this.peer = peer;

        peer.ontrack = e => {
            this.mediaStream = e.streams[0];
        };

        peer.onicecandidate = e => {
            if (e.candidate) {
                this.sendMessage(MessageHelpers.createMessage(Messages.iceCandidate, {
                    candidate: e.candidate
                }));
            }
        };

        const sdpOffer: RTCSessionDescriptionInit = {
            sdp: message.sdp,
            type: 'offer'
        };

        await peer.setRemoteDescription(new wrtc.RTCSessionDescription(sdpOffer));
        const answer = await peer.createAnswer();
        await peer.setLocalDescription(answer);

        this.sendMessage(MessageHelpers.createMessage(Messages.iceCandidate, { sdp: answer.sdp }));
        this.connectionPromise?.resolve(peer);
    }

[Igrium]

Apparently, in the context of a browser, it has something to do with it not liking the codec, but that doesn't really make sense when I'm not even attempting to decode it server-side.

[duvallj]

Hi @Igrium,

The error message seems to come from https://webrtc.googlesource.com/src.git/+/a6b138d6b4ef3a5b2c87f899b67f3b5c8dd3c002/pc/channel.cc#1120, with the implementation at https://webrtc.googlesource.com/src.git/+/a6b138d6b4ef3a5b2c87f899b67f3b5c8dd3c002/media/engine/webrtc_video_engine.cc#864, which seems to validate the hypothesis that libwebrtc doesn't like the codec/something else about the SDP, since something in GetChangedSendParameters is failing.

This is a fairly old version of libwebrtc, so it may make sense that there's something in here Unreal Engine is trying to negotiate that's too new for it to understand:

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=42e01f

H264/90000 is something everyone knows, same with nack/nack pli, but the rest i am not so sure. Also could be something from a different section who knows!

In any case, more investigation is required. If you'd like, I can provide you with a debug build w/ more logging, might help us narrow down what part is problematic at least. What platform do you use?

[Igrium]

Yeah, that would be great. I'm using Windows for development, although I intend to use Docker in production (on linux)

[Igrium]

How would I install it? Given that I'm currently using the build from NPM

[duvallj]

I don't have any debug binaries published on NPM unfortunately; you'll have to modify your environments directly somehow.

I've manually published the just debug binary I've built from source on my local machine to https://static.duvallj.pw/node-webrtc/v0.9.0/debug-win32-x64/wrtc.node; you can find a linux version at the corresponding debug-linux-x64 & debug-linux-arm64 URLs. What you'll want to do then is, in your node_modules folder, replace the wrtc.node file you find there with the one you downloaded from this URL.

Chrome will probably flag this as a suspicious download, and for good reason! It's unsigned code downloaded from the internet! I don't think I would trust me here, may be safer building from source but that's also a lot more work. Up to you.

[Igrium]

Now it just does this:

/home/igrium/webrovers-server/node_modules/@roamhq/wrtc/lib/binding.js:28
  throw new Error(
        ^
Error: Could not find wrtc binary on any of the paths: ../build-linux-x64/wrtc.node,../build-linux-x64/Debug/wrtc.node,../build-linux-x64/Release/wrtc.node,@roamhq/wrtc-linux-x64,./node_modules/@roamhq/wrtc-linux-x64,./node_modules/@roamhq/wrtc-linux-x64/wrtc.node
    at Object.<anonymous> (/home/igrium/webrovers-server/node_modules/@roamhq/wrtc/lib/binding.js:28:9)
    at Module._compile (node:internal/modules/cjs/loader:1730:14)
    at node:internal/modules/cjs/loader:1895:10
    at Object.require.extensions.<computed> [as .js] (/home/igrium/webrovers-server/node_modules/ts-node/src/index.ts:1608:43)
    at Module.load (node:internal/modules/cjs/loader:1465:32)
    at Function._load (node:internal/modules/cjs/loader:1282:12)
    at TracingChannel.traceSync (node:diagnostics_channel:322:14)
    at wrapModuleLoad (node:internal/modules/cjs/loader:235:24)
    at Module.require (node:internal/modules/cjs/loader:1487:12)
    at require (node:internal/modules/helpers:135:16)
[nodemon] app crashed - waiting for file changes before starting...

[duvallj]

Ah, that's probably my bad; I build these Linux ones on NixOS, which is well-known for generating binaries that do not port to any other computer. I think I know what caused it (accidentally skipped a step in the process (patchelf --remove-rpath wrtc.node)), and have updated the binary at https://static.duvallj.pw/node-webrtc/v0.9.0/debug-linux-x64/wrtc.node, do you mind giving it another go?

[Igrium]

Okay I have no idea what any of this means, but it gave me this:

Signaling server started on port 8888
New streamer connection:  ::ffff:192.168.160.1
StreamerRegistry: Streamer ID change: UnknownStreamer -> Editor
Attempting connection to peer.
(audio_processing_impl.cc:282): Injected APM submodules:
Echo control factory: 0
Echo detector: 0
Capture analyzer: 0
Capture post processor: 0
Render pre processor: 0
(audio_processing_impl.cc:291): Denormal disabler: supported
(webrtc_voice_engine.cc:321): WebRtcVoiceEngine::WebRtcVoiceEngine
(webrtc_video_engine.cc:613): WebRtcVideoEngine::WebRtcVideoEngine()
(webrtc_voice_engine.cc:343): WebRtcVoiceEngine::Init
(webrtc_voice_engine.cc:439): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, typing: 1, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 1, }
(webrtc_voice_engine.cc:548): Stereo swapping enabled? 0
(webrtc_voice_engine.cc:553): NetEq capacity is 200
(webrtc_voice_engine.cc:559): NetEq fast mode? 0
(webrtc_voice_engine.cc:565): NetEq minimum delay is 0
(webrtc_voice_engine.cc:571): NetEq handle reordered packets? 0
(webrtc_voice_engine.cc:637): NS set to 1
(webrtc_voice_engine.cc:641): Typing detection is enabled? 1
(audio_processing_impl.cc:523): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_level_step: 15, clipped_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor:  { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3, use_predicted_step: 1 }}}, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, headroom_db: 6, max_gain_db: 30, initial_gain_db: 8, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50}}, residual_echo_detector: { enabled: 1 }}
(transparent_mode.cc:239): AEC3 Transparent Mode: Legacy
(echo_canceller3.cc:750): AEC3 created with sample rate: 16000 Hz, num render channels: 1, num capture channels: 1
(clipping_predictor.cc:357): [agc] Clipping prediction disabled.
(agc_manager_direct.cc:69): [agc] GetMinMicLevel
(agc_manager_direct.cc:73): [agc] Using default min mic level: 12
(agc_manager_direct.cc:520): AgcManagerDirect::Initialize
(peer_connection_factory.cc:344): Using default network controller factory
(bitrate_prober.cc:72): Bandwidth probing enabled, set to inactive
(cpu_info.cc:53): Available number of cores: 12
(aimd_rate_control.cc:112): Using aimd rate control with back off factor 0.85
(remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating.
(remote_estimator_proxy.cc:47): Maximum interval between transport feedback RTCP messages (ms): 250
(openssl_key_pair.cc:38): Making key pair
(openssl_key_pair.cc:91): Returning key pair
(boringssl_certificate.cc:187): Making certificate for WebRTC
(boringssl_certificate.cc:243): Returning certificate
(p2p_transport_channel.cc:586): Set backup connection ping interval to 25000 milliseconds.
(p2p_transport_channel.cc:595): Set ICE receiving timeout to 2500 milliseconds
(p2p_transport_channel.cc:602): Set ping most likely connection to 0
(p2p_transport_channel.cc:609): Set stable_writable_connection_ping_interval to 2500
(p2p_transport_channel.cc:622): Set presume writable when fully relayed to 0
(p2p_transport_channel.cc:640): Set regather_on_failed_networks_interval to 300000
(p2p_transport_channel.cc:647): Set receiving_switching_delay to 1000
(jsep_transport_controller.cc:1058): Creating DtlsSrtpTransport.
(dtls_srtp_transport.cc:63): Setting RTCP Transport on 0 transport 0
(dtls_srtp_transport.cc:68): Setting RTP Transport on 0 transport 60002280
(p2p_transport_channel.cc:586): Set backup connection ping interval to 25000 milliseconds.
(p2p_transport_channel.cc:595): Set ICE receiving timeout to 2500 milliseconds
(p2p_transport_channel.cc:602): Set ping most likely connection to 0
(p2p_transport_channel.cc:609): Set stable_writable_connection_ping_interval to 2500
(p2p_transport_channel.cc:622): Set presume writable when fully relayed to 0
(p2p_transport_channel.cc:640): Set regather_on_failed_networks_interval to 300000
(p2p_transport_channel.cc:647): Set receiving_switching_delay to 1000
(jsep_transport_controller.cc:1058): Creating DtlsSrtpTransport.
(dtls_srtp_transport.cc:63): Setting RTCP Transport on 1 transport 0
(dtls_srtp_transport.cc:68): Setting RTP Transport on 1 transport 60079180
(p2p_transport_channel.cc:586): Set backup connection ping interval to 25000 milliseconds.
(p2p_transport_channel.cc:595): Set ICE receiving timeout to 2500 milliseconds
(p2p_transport_channel.cc:602): Set ping most likely connection to 0
(p2p_transport_channel.cc:609): Set stable_writable_connection_ping_interval to 2500
(p2p_transport_channel.cc:622): Set presume writable when fully relayed to 0
(p2p_transport_channel.cc:640): Set regather_on_failed_networks_interval to 300000
(p2p_transport_channel.cc:647): Set receiving_switching_delay to 1000
(jsep_transport_controller.cc:1058): Creating DtlsSrtpTransport.
(dtls_srtp_transport.cc:63): Setting RTCP Transport on 2 transport 0
(dtls_srtp_transport.cc:68): Setting RTP Transport on 2 transport 6007b840
(p2p_transport_channel.cc:531): Received remote ICE parameters: ufrag=Rr19, renomination disabled
(p2p_transport_channel.cc:531): Received remote ICE parameters: ufrag=Rr19, renomination disabled
(p2p_transport_channel.cc:531): Received remote ICE parameters: ufrag=Rr19, renomination disabled
(sdp_offer_answer.cc:3284): Adding video transceiver for MID=0 at i=0 in response to the remote description.
(webrtc_video_engine.cc:626): CreateMediaChannel. Options: VideoOptions {}
(channel.cc:137): Created channel: {mid: 0, media_type: video}
(rtp_demuxer.cc:145): Added sink = 6007f418 for criteria {mid: 0, rsid: <empty>, ssrcs: [], payload_types = []}
(rtp_transceiver.cc:168): Blocking SetChannel: total=2 (actual=2, could=0)
(sdp_offer_answer.cc:3284): Adding audio transceiver for MID=1 at i=1 in response to the remote description.
(webrtc_voice_engine.cc:1558): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
(webrtc_voice_engine.cc:439): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
(webrtc_voice_engine.cc:553): NetEq capacity is 200
(webrtc_voice_engine.cc:559): NetEq fast mode? 0
(webrtc_voice_engine.cc:565): NetEq minimum delay is 0
(webrtc_voice_engine.cc:571): NetEq handle reordered packets? 0
(audio_processing_impl.cc:523): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_level_step: 15, clipped_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor:  { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3, use_predicted_step: 1 }}}, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, dry_run: 0, headroom_db: 6, max_gain_db: 30, initial_gain_db: 8, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50}}, residual_echo_detector: { enabled: 1 }}
(webrtc_voice_engine.cc:1576): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, }
(channel.cc:137): Created channel: {mid: 1, media_type: audio}
(rtp_demuxer.cc:145): Added sink = 60081118 for criteria {mid: 1, rsid: <empty>, ssrcs: [], payload_types = []}
(rtp_transceiver.cc:168): Blocking SetChannel: total=2 (actual=2, could=0)
(sdp_offer_answer.cc:3414): Creating data channel, mid=2
(peer_connection.cc:2410): Setting up data channel transport for mid=2
(sdp_offer_answer.cc:2470): Session: 1232405551961500850 Old state: stable New state: have-remote-offer
(webrtc_video_engine.cc:1574): ResetUnsignaledRecvStream.
(webrtc_voice_engine.cc:2047): ResetUnsignaledRecvStream.
(channel.cc:1080): Setting remote video description for {mid: 0, media_type: video}
(webrtc_video_engine.cc:867): SetSendParameters: {codecs: [VideoCodec[127:H264], VideoCodec[97:rtx], VideoCodec[98:H264], VideoCodec[99:rtx], VideoCodec[100:red], VideoCodec[101:rtx], VideoCodec[102:ulpfec]], conference_mode: no, extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 8}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}], extmap-allow-mixed: true, max_bandwidth_bps: -1, mid: 0}
(webrtc_video_engine.cc:804): No video codecs supported.
(sdp_offer_answer.cc:4275): Failed to set remote video description send parameters for m-section with mid='0'. (INVALID_PARAMETER)
(sdp_offer_answer.cc:2197): Failed to set remote offer sdp: Failed to set remote video description send parameters for m-section with mid='0'.
[DOMException [InvalidAccessError]: Failed to set remote offer sdp: Failed to set remote video description send parameters for m-section with mid='0'.]

[duvallj]

Those logs help a lot, thanks! The relevant lines are this:

(webrtc_video_engine.cc:867): SetSendParameters: {codecs: [VideoCodec[127:H264], VideoCodec[97:rtx], VideoCodec[98:H264], VideoCodec[99:rtx], VideoCodec[100:red], VideoCodec[101:rtx], VideoCodec[102:ulpfec]], conference_mode: no, extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:3gpp:video-orientation, id: 3}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 4}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 8}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 10}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 11}], extmap-allow-mixed: true, max_bandwidth_bps: -1, mid: 0}
(webrtc_video_engine.cc:804): No video codecs supported.

These help explain exactly why it's failing. Requires more tracing back from https://webrtc.googlesource.com/src.git/+/a6b138d6b4ef3a5b2c87f899b67f3b5c8dd3c002/media/engine/webrtc_video_engine.cc#800 to figure out why, but thanks for doing this! I'll try to get around to this eventually.

[Igrium]

In the mean time, what should I do about it?