WebRTC: Failed to parse video codecs correctly.
champsupertramp opened this issue · 6 comments
Why do i keep getting this error with your webrtc-examples in Chrome and Safari browser? Sometimes i had to click the "Start streaming" multiple times and then it will work. I'm using the latest WebRTC from this repo.
Here's the video source and transcoder settings:
Live Stream Type
Adaptive bitrate
Billing Mode
Pay as you go
Delivery Protocol
Apple HLS
Reduced Latency Stream
No
Source Aspect Ratio
16:9 (Widescreen) -- 640 x 360
Recording
Yes
Closed Captions
None
Your Wowza WebRTC publisher also throws this error. I'm using the latest Safari and Chrome browsers in MacOS.
Wowza WebRTC Publisher logs:
[Log] Sdp Data: v=0
(index.bundle.k98vxqwb.js, line 1)
o=WowzaStreamingEngine-next 777736795 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=msid-semantic: WMS *
m=audio 9 UDP/TLS/RTP/SAVPF 111
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:9f272078
a=ice-pwd:1b98c37434c20ffa66a8873147152157
a=ice-options:trickle
a=fingerprint:sha-256 C6:43:59:4F:77:FB:E1:D5:32:35:F0:C8:88:E0:16:62:C7:33:0C:2A:0E:11:30:F9:B7:F7:BB:B5:DC:DA:32:F1
a=setup:passive
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=recvonly
a=msid:2acfd567-a23e-4b68-8cee-004aea025a68 c048e845-7f67-4052-a97a-c8b8d1cf9c50
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 x-google-min-bitrate=NaN;x-google-max-bitrate=NaN
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:111 opus/48000/2
a=fmtp:111 x-google-min-bitrate=NaN;x-google-max-bitrate=NaN
a=fmtp:111 x-google-min-bitrate=64;x-google-max-bitrate=64
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
m=video 9 UDP/TLS/RTP/SAVPF 98
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:9f272078
a=ice-pwd:1b98c37434c20ffa66a8873147152157
a=ice-options:trickle
a=fingerprint:sha-256 C6:43:59:4F:77:FB:E1:D5:32:35:F0:C8:88:E0:16:62:C7:33:0C:2A:0E:11:30:F9:B7:F7:BB:B5:DC:DA:32:F1
a=setup:passive
a=mid:1
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:10 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=recvonly
a=msid:2acfd567-a23e-4b68-8cee-004aea025a68 8f25fd8b-706c-47f5-8acc-872e746be424
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:98 H264/90000
a=fmtp:98 x-google-min-bitrate=NaN;x-google-max-bitrate=NaN
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=42e01f
a=rtpmap:98 H264/90000
a=fmtp:98 x-google-min-bitrate=NaN;x-google-max-bitrate=NaN
a=fmtp:98 x-google-min-bitrate=750;x-google-max-bitrate=750
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=42e01f
Are you using Wowza Streaming Engine or Cloud? Configured for UDP or TCP? I have not had this issue, I usually run in Chrome on MacOS with Wowza Streaming Engine.
Your Wowza WebRTC publisher also throws this error. I'm using the latest Safari and Chrome browsers in MacOS.
This is not part of our repo examples.
Hi @akeller I'm using the Wowza Cloud Service.
Regards,
This is not part of our repo examples today and is a different code base than the ones available here. Please submit a support ticket for further debugging as I am not able to replicate the behavior you are seeing when using Wowza Streaming Cloud with the Publish page in Chrome.
I will also respond to your forum post as well.