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WebRTC firefox[47.0.1] client lowers the bitrate to very low values sometimes till 0.02 Mbps.

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  • 최종 답변자: Standard8

mzak
mzak
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Hello,

Outgoing H.264 call from webRTC Firefox client version 47.0.1. We use the H.264 codec with packetization-mode=0 version.

We observe strange behavior of the Firefox browser - it lowers the bitrate of 

the outgoing call to very low values of the bitrate sometimes till 0.02 Mbps.

We do not observe packet loss, so it could not be the reason to lower the bitrate.

The video section [excluding ICE/TURN/STUN/SRTP stuff] of our SDP is below:

m=video 60020 RTP/AVP 108 c=IN IP4 31.168.3.180 b=TIAS:1200000 a=rtpmap:108 H264/90000 a=fmtp:108 profile-level-id=42801f; max-fs=3600; packetization-mode=0 a=content:main a=label:1 a=rtcp-fb:* ccm fir a=rtcp-fb:* nack pli a=sendrecv

We cannot explain the reason why firefox client lowers the bitrate. The issue does not happen with chrome 52 beta- supporting H.264. Do you have any suggestion what could be the reason of the lowering bit-rate?

Thanks in advance Michael

Hello, Outgoing H.264 call from webRTC Firefox client version 47.0.1. We use the H.264 codec with packetization-mode=0 version. We observe strange behavior of the Firefox browser - it lowers the bitrate of the outgoing call to very low values of the bitrate sometimes till 0.02 Mbps. We do not observe packet loss, so it could not be the reason to lower the bitrate. The video section [excluding ICE/TURN/STUN/SRTP stuff] of our SDP is below: m=video 60020 RTP/AVP 108 c=IN IP4 31.168.3.180 b=TIAS:1200000 a=rtpmap:108 H264/90000 a=fmtp:108 profile-level-id=42801f; max-fs=3600; packetization-mode=0 a=content:main a=label:1 a=rtcp-fb:* ccm fir a=rtcp-fb:* nack pli a=sendrecv We cannot explain the reason why firefox client lowers the bitrate. The issue does not happen with chrome 52 beta- supporting H.264. Do you have any suggestion what could be the reason of the lowering bit-rate? Thanks in advance Michael

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mzak
mzak 질문자
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Hi,

We checked: when we work with packetization-mode=1 - everything works perfect. Apparently the problem exists only in single NALU mode, I recommend not to use packetization-mode=0 at all in SDP - if it does not work good.

Best Regards Michael

Standard8
Standard8
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Hi, I would suggest posting this to the dev.media mailing list:

https://lists.mozilla.org/listinfo/dev-media

That's where most of our WebRTC experts hang out and should be able to provide you with a good response.