Hi, I’m trying to use PlainRtpTransport for ingesting multiple RTP flows from a GStreamer pipeline in order to obtain a Simulcast producer using VP8 codec. I changed the “encodings” array in the rtpParameters structure, adding 3 levels setting scalabilityMode = “S1T1” and adding 3 different ssrc v…

OK OK I see! And actually it’s nice that you are trying this. So: I thought your consumer was not using REMB, my fault. With current just-REMB BWE in mediasoup v3 and the current probation system (based on real video packet retransmissions) your scenario is not possible. Actually you can se…

Yes, I see from conference streaming :slight_smile: More investigation in progress.

Everything seems ok: RTCP packets are sent and received. Can you anyway check SimulcastConsumer::CanSwitchToSpatialLayer() to see whether it returns true or false? The only problem seems that the DistributeAvailableOutgoingBitrate method doesn’t evaluate correctly the availableBitrate. If REMB…

[image] ibc: DistributeAvailableOutgoingBitrate() is not called if there is no REMB support The client is a web browser, so the consumer has REMB support, right? I configured the simulcast bitrates in the rtp producer as follows: 'encodings': [ {'ssrc': 1000, 'active': True, 'maxBitrate':…

Adding some debug messages, I found that in my case (3 spatial and 1 temporal layers), this loop exits here ( https://github.com/versatica/mediasoup/blob/5c2208d5e70324ef28a91106afdeb67330ac70bd/worker/src/RTC/SimulcastConsumer.cpp#L443 ): if (requiredBitrate > virtualBitrate) goto done; …

[image] ibc: Actually you can set the initialOutgoingAvailableBitrate in the consumer transport (which defaults to 600000 bps) to a higher value, something higher than the low stream you are producing (2 mbps). That will let the low stream reach the receiver browser. Maybe initialAvailableOut…

[image] ibc: And may be we’ll implement NADA instead of trying to figure out the algorithm of libwebrtc You can find a comparison here: Med-conf-gcc.pdf 776.47 KB

[image] vpalmisano: Maybe initialAvailableOutgoingBitrate. Setting this option the problem is resolved. [image] ibc: But it’s not perfect. You may not be able to up switch to the next layer later. [image] vpalmisano: I think that using simulcast, at least the smaller level should be sen…

Oh cool! something to read during my flight tomorrow :slight_smile: Beside it, do you think NADA is the good approach? Well, GCC algorithm is not documented so…

[image] ibc: Beside it, do you think NADA is the good approach? Luca (the paper author) says that NADA (at the time of the publication) had some queueing problems. He doesn’t know if these problems are fixed in the latest release.

[image] vpalmisano: Maybe initialAvailableOutgoingBitrate. Setting this option the problem is resolved. BTW, until we have proper sender side BWE done, you can use an undocumented parameter in router.createWebRtcTransport(): * @param {Number} [initialAvailableOutgoingBitrate=600000] - Initia…

Using PlainRtpTransport for Simulcast ingesting

mediasoup libraries

saket424 (Saket424) December 2, 2019, 2:47pm 29

@vpalmisano,
Could you share with us the gstreamer video simulcast example. Thanks in advance

Using PlainRtpTransport for simulcast ingesting

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Using PlainRtpTransport for Simulcast ingesting

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