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They were once added to be used for audio bandwidth estimation. We decided to not pursue that solution and have therefore removed abs-send-time for audio in Chrome M56.For synchronizing streams you should be using RTCP sender reports./Stefan
On Mon, Nov 28, 2016 at 8:10 AM Tian Yong <hass...@gmail.com> wrote:
HeyWhen we recording in our webrtc gateway, sometimes it went out of sync, it very difficult for us debug this, so we use the abs-send-time in audio(opus) rtp headers, it works very good.So is this abs-send-time used for? or there's some other use?Cheers!
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We have never included audio in remb, that never launched. For audio BWE we recommend using send-side BWE which is default in M55 and uses these extensions:
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
Maybe you don't include audio in REMB calculation in M56?What do you use instead of abs-send-time in Chrome M56?Receiver side REMB seems to rely on the abs-send-time extension.On Tue, Nov 29, 2016 at 11:37 AM, 'Stefan Holmer' via discuss-webrtc <discuss...@googlegroups.com> wrote:
They were once added to be used for audio bandwidth estimation. We decided to not pursue that solution and have therefore removed abs-send-time for audio in Chrome M56.For synchronizing streams you should be using RTCP sender reports./Stefan
On Mon, Nov 28, 2016 at 8:10 AM Tian Yong <hass...@gmail.com> wrote:
HeyWhen we recording in our webrtc gateway, sometimes it went out of sync, it very difficult for us debug this, so we use the abs-send-time in audio(opus) rtp headers, it works very good.So is this abs-send-time used for? or there's some other use?Cheers!
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We have never included audio in remb, that never launched. For audio BWE we recommend using send-side BWE which is default in M55 and uses these extensions:
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=rtcp-fb:100 transport-ccNote that audio BWE is experimental and not enabled by default./Stefan
On Tue, Nov 29, 2016, 11:34 pablo platt <pablo...@gmail.com> wrote:
Maybe you don't include audio in REMB calculation in M56?What do you use instead of abs-send-time in Chrome M56?Receiver side REMB seems to rely on the abs-send-time extension.
On Tue, Nov 29, 2016 at 11:37 AM, 'Stefan Holmer' via discuss-webrtc <discuss-webrtc@googlegroups.com> wrote:
They were once added to be used for audio bandwidth estimation. We decided to not pursue that solution and have therefore removed abs-send-time for audio in Chrome M56.For synchronizing streams you should be using RTCP sender reports./Stefan
On Mon, Nov 28, 2016 at 8:10 AM Tian Yong <hass...@gmail.com> wrote:
HeyWhen we recording in our webrtc gateway, sometimes it went out of sync, it very difficult for us debug this, so we use the abs-send-time in audio(opus) rtp headers, it works very good.So is this abs-send-time used for? or there's some other use?Cheers!
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Why audio wasn't included in REMB?I'm including it in my implementation and wonder if I should ignore audio packets.
Where is the send-side BWE code?
I can see REMB but I can't find send-side.
https://chromium.googlesource.com/external/webrtc/+/master/webrtc/modules/remote_bitrate_estimator/On Wed, Nov 30, 2016 at 9:46 AM, 'Stefan Holmer' via discuss-webrtc <discuss...@googlegroups.com> wrote:
We have never included audio in remb, that never launched. For audio BWE we recommend using send-side BWE which is default in M55 and uses these extensions:
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=rtcp-fb:100 transport-ccNote that audio BWE is experimental and not enabled by default./Stefan
On Tue, Nov 29, 2016, 11:34 pablo platt <pablo...@gmail.com> wrote:
Maybe you don't include audio in REMB calculation in M56?What do you use instead of abs-send-time in Chrome M56?Receiver side REMB seems to rely on the abs-send-time extension.
On Tue, Nov 29, 2016 at 11:37 AM, 'Stefan Holmer' via discuss-webrtc <discuss...@googlegroups.com> wrote:
They were once added to be used for audio bandwidth estimation. We decided to not pursue that solution and have therefore removed abs-send-time for audio in Chrome M56.For synchronizing streams you should be using RTCP sender reports./Stefan
On Mon, Nov 28, 2016 at 8:10 AM Tian Yong <hass...@gmail.com> wrote:
HeyWhen we recording in our webrtc gateway, sometimes it went out of sync, it very difficult for us debug this, so we use the abs-send-time in audio(opus) rtp headers, it works very good.So is this abs-send-time used for? or there's some other use?Cheers!
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http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
transport-cc
Thanks
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Hello Stefan,
first of all thanks for clarifying this.
Anyway I don't know if I am missing something related to the send-side BWE, but I think that both audio and video RTP packets should be taken into account, because there can be several audio streams that can significantly affect the network bandwidth (and even in an ideal case DataChannels should be considered).
Hence, why do you decide to remove abs-send-time for audio?
If I am not wrong, both abs-send-time and transport-cc RTP header extensions are needed for the send-side BWE algorithm, aren't they?
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