Playign with googAvailableReceiveBandwidth

[Piyush Ranjan]

Hi

I am trying to better the quality of the stream by increasing the limit of bandwidth webrtc can use. I have seen this, googAvailableReceiveBandwidth and googAvailableSendBandwidth, in getStats. The value that it auto discovers is way low than what I have available between peers. For instance, even on my local network it settles at 250kbps whereas I have 100mbps switch.

I was wondering if there was a way to play with this value ? So that I can provide this value as what it actually available bandwidth between peers so that stream is of better quality. I control both the peers so I am not averse to recompiling and using custom chrome/node-webrtc peers. 

Thanks

Piyush

[Vikas]

Hi,

 You can try b=AS sdp attribute to set maximum value of audio/video bitrate but i think you should also check if you have packet loss/delay in your network that is causing the bandwidth estimation to not work reliably.

/Vikas

[Justin Uberti]

Note also that WebRTC will not estimate bandwidth above what it is actually using. (i.e. don't expect it to report 100 mbps when sending a 360p stream)

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[Piyush Ranjan]

Vikas

I have tried this with chrome 34 beta. The tab on which the rtc is running crashes after offer is answered. I am looking into if I am doing something wrong in composition. 

Justin

Thank you for this reply. There are packet drops but not more than 2% of overall packets. I think that should be acceptable as I am doing this over internet. 

I am using getStats to print information on page and update every second. The problem, I am facing, is that available bandwidth goes up to 1500kbps but when there is some activity (when I browse through a heavy page) on the 'server' side then the available bandwidth drop precipitously to even 30kbps. This results in very bad quality of the screencapture presented to user. The image slowly improves as soon as page is loaded and cpu usage drops. I need quality to remain good even during page transitions. The bandwidth between presenting machine 'server' and client machine is more than 5000kbps at any point of time.

[Emiliano delli Carri]

Hello to all. 

I'm testing SipML5 + webrtc2sip + Asterisk. 

I have set the parameter b = AS as described in seguante link:

https://groups.google.com/forum/#!searchin/doubango/bandwidth/doubango/WJOF3Y8qVGc/UnDHD2xjicYJ 

But googAvailableReceiveBandwidth (as you can see from the attached) tends to grow up to 2 Mbps and the video freezes. How can I limit the bandwidth in reception?

Please help me. 

Thank you.

Emiliano

[Darshan Shankar]

Piyush,

Did you make any progress on this? 

I'm trying to improve the quality of the stream as well, especially in the beginning of the stream. I control both ends, and have been compiling Chrome from source and modifying the video engine parameters to use larger initial bandwidth values, etc. 

I'm curious if you (or anyone else) has made progress on improving Chrome's bandwidth estimation.

Why isn't WebRTC using more bandwidth on a 1080p stream? Wish there was a way to improve quality and use more bandwidth.

[Lorenzo Miniero]

Emiliano,

that's likely an issue with RTCP packets, that are either discarded (Asterisk doesn't handle REMB) or have incorrect information in it (e.g., wrong SSRC, see http://tools.ietf.org/html/draft-miniero-straw-b2bua-rtcp-00 for some more info).

Lorenzo