Webrtc Turn Config3/21/2021
To function correctly, it must be reachable by both the external devices.An Expressway-E, acting as a TURN server, will bridge the traffic.
This is an Expressway X8.11 and later feature that allows the external clients to use port 443 to access the TURN. Note: This may be the default Chrome home page for some clients. If you like, you can now continue on to add support for Skype for. We have multiple other optional lab sections, depending on your time. Do you prefer the alternative of having no connection to having a connection that isnt the best it could be. What I didnt cover in that article was how TURN over TCP changes the behavior we end up seeing on the network. This is going to be AppRTC, 1:1 call, with no network impairments and no use of TURN whatsoever. We will focus on the video stats, because theres a lot more data in them. With longer calls, it would average at 2.5Mbps in both directions. ![]() I forced TCP relay by blocking all UDP traffic in our machines. Notice how the outgoing video bitrate ramps up a lot faster than the incoming video bitrate Two reasons why this might be happening. When UDP is used, WebRTC is a lot more agressive (and accurate) about estimating the available bitrate. So on the outgoing, WebRTC estimates that theres enough bitrate to use, but then on the incoming, TCP slows everything down, ramping up to 2.4Mbps in 30 seconds instead of less than 5 that were used to by WebRTC. Webrtc Turn Config Full 1Mbps ThatThat interesting hump we have for the video, where we have a jump of almost a full 1Mbps that goes back down later That hump also coincides with packet loss reporting at the beginning of it something that is weird as well remember that TCP doesnt lose packets it re-transmits them. Were now at over 3Mbps for the same type of content because of 0.5 packet loss. WebRTC saw the opportunity to pump more bits to deal with the network and so it did. And since we didnt really limit it in this test it took the right approach. Which brings me to the other thing the packet loss chart seems especially clean. Thats because TCP hides that and re-transmit everything so as not to lose packets. It also means that we have utilization of bitrate that is way higher than the 1.9Mbps it is just not available for WebRTC and in most cases, these re-tramsnissions dont really help WebRTC at all as they come too late to play them back anyway. This is why Ive had more success on 3G links with a WebRTC call than on corporate wireless networks funny it should have come to that. What is the best way to simulate a shoddy WIFI with a strict firewall Appreciate your response. Tsahi described this over at TestRTC a while ago, showing the impact on bitrate and other. To my knowledge, WebRTC, like SIP and H.323, is meant to work using UDP. UDP is the protocol suitable for realtime data, whereas TCP is slow and uses acknowledgements. It is better to discard a video packet than to send it later, there is no use for it. I have not found any place where WebRTC is defined (I guess Google can only define it) as over TCP. I even think of saying This is the wrong setup, WebRTC is defined using UDP or even saying you are not compliant to the WebRTC specification. What do you think about the above Do you find it wrong like me or do you see it as an acceptable practice.
0 Comments
Leave a Reply.AuthorWrite something about yourself. No need to be fancy, just an overview. ArchivesCategories |