bitrate constraints for rtc data channel?

[Bill Gibson]

My JS Chrome app is trying to push a good bit of data to other peers. The bit rate when passing to local/LAN peers is pretty fast, but if the peers are not local the rate drops well below max upload speed (10x maybe more). I can't use the getStats on a data channel, so that isn't something I can use to review performance. For example, for a >1Mbit/second upload speed connection, I am only able to push ~120Kbit/second. For my application, this is too slow. The media streams seem to get better performance, or am I wrong?

What kind of flow control is currently enabled for data channel connections? Is it possible to tune this or other ways to push more bits through?

Thank you,

Bill

[Benjamin Schwartz]

The data channel uses SCTP, which includes automatic flow control with congestion avoidance, similar to TCP.  In general, this should be able to use most of the available bandwidth on the link.

You should check the packet loss rate on the link, because SCTP (like TCP) reduces bandwidth when there is packet loss, whereas the media streams (using RTP) are much more tolerant of packet loss.

You should also check the ping time on the link.  If it is high, you might be encountering a known issue in Chrome:

https://code.google.com/p/webrtc/issues/detail?id=3695 .  You might want to compare with Firefox on the same link.

--

---
You received this message because you are subscribed to the Google Groups "discuss-webrtc" group.
To unsubscribe from this group and stop receiving emails from it, send an email to discuss-webrt...@googlegroups.com.
For more options, visit https://groups.google.com/d/optout.