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Fix Rtp padding generator for older firefox build. #1152
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Also set marker to false per RFC
does these changes still works for Chrome? I think to remember we had some issues in the past with the marker set to true (but I might be wrong). |
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I like this approach! I just wonder if it could affect the way the padding works for Chrome. I'll try to do some tests in my local environment during the next week. Have you tested it already for Chrome too?
@@ -100,6 +100,7 @@ void RtpPaddingGeneratorHandler::sendPaddingPacket(std::shared_ptr<DataPacket> p | |||
RtpHeader *rtp_header = reinterpret_cast<RtpHeader*>(padding_packet->data); | |||
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rtp_header->setSeqNumber(sequence_number.output); | |||
rtp_header->setMarker(false); |
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could we set the marker bit inside RtpUtils::makePaddingPacket please?
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I'm sorry idk why i messed up here
sendPaddingPacket(packet, last_padding_packet_size_); | ||
// Temporary fix since https://bugzilla.mozilla.org/show_bug.cgi?id=1435025 is fixed | ||
// Only send full rtp padding packet as suggested also in webrtc code. | ||
// sendPaddingPacket(packet, last_padding_packet_size_); |
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this could affect the way the congestion control works, have you tested it with Chrome too? did you notice any significant change in the amount of time it takes for simulcast to increase the quality layers?
I tested with chrome and seems ok! also because chrome itself send the packet with the marker false. Also, I don't think that the minimum padding size that we drop here will affect much the increase rate of bw estimation (but I could be wrong). If you can also test this will be great, but I don't think this will cause any issue. |
Also set marker to false per RFC
Idk if it needs some unit tests
[] It needs and includes Unit Tests
[] It includes documentation for these changes in
/doc
.