<div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Feb 17, 2015 at 9:24 AM, Nitesh Bansal <span dir="ltr"><<a href="mailto:nitesh.bansal@gmail.com" target="_blank">nitesh.bansal@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div>Hi Matt,<br><br></div>It seems that my reply was lost in the pile of mails during holiday period?<br><br></div><div>Is there any update on this?<br><br></div><div>Regards,<br></div><div>Nitesh<br></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Dec 30, 2014 at 5:54 PM, Nitesh Bansal <span dir="ltr"><<a href="mailto:nitesh.bansal@gmail.com" target="_blank">nitesh.bansal@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div>Hi Matt,<br><br></div>The scenario which i have in my mind is as follows:<br><br></div>WebRTC browser -----------> (Inbound) Asterisk --------------> SIP phone (VP.8 video call)<br><br></div>Since this is a webRTC call, RTP stream will be encrypted between Web Browser and Asterisk (call leg 1) and in our case and call leg between Asterisk and SIP phone(call leg 2) can be secured/unsecured depending on the config. Since one leg of the call is encrypted, there can't be a P2P bridge.<br></div><div>But the fact is that asterisk isn't the source of the video, it is just forwarding the RTP stream.<br></div><div>In this case, i believe it makes sense to forward RTCP FB messages like NACK, PLI and FIR.<br></div><div>NACKs -- Asterisk can request the retransmission of video frames <br></div><div>PLI/FIR -- Asterisk can request image refresh from either endpoint<br><br></div><div>To avoid the RTCP FB forwarding in other scenarios, we can probably make it as config enabled feature.<br></div><div><br></div></div></blockquote></div></div></div></div></blockquote><div><br></div><div>It wasn't really lost in the pile. I'm just not sure I have anything else to say that I haven't already said.<br><br>It isn't that your idea won't work - for your specific scenario, it will absolutely work. But I can't see how it universally applicable, and the code is going to be intrusive enough that I'm not sure I'd want a non-universal solution.<br><br></div><div>In your outlined scenario, Asterisk is not forwarding the media. It may seem like it - but it isn't. It is taking the RTP packet, putting it into an AST_FRAME_VIDEO (along with AST_FRAME_VOICE most likely as well), passing them over a bridge - which may go to more than one party - re-packaging it in RTP, and passing it along. Yes, it isn't transcoding it, but that doesn't mean that there aren't scenarios where from one party's perspective, the media from Asterisk has remained the same, while in reality some other party is now sending it media.<br>should be forwarded might help here. Right now, I'm having a hard time envisioning the specific scenario where you have two or more endpoints providing video to Asterisk and where Asterisk can simply relay the entire feedback message and not generate the message itself. I could be mistaken, but it feels like while this could work in a simple two-party core bridge, there are a lot of other scenarios where this goes wrong quickly.<br></div><br></div>I could be wrong, and there may be a graceful way to support this in your scenario that doesn't violate Asterisk's model of channels and acting as a B2BUA. But I suspect that multi-party bridges, call hold, redirecting channels in/out of bridges, Local channels, and more will cause lots of problems for the implementation that you have in mind.<br><br>Matt<br clear="all"></div><div class="gmail_extra"><br>-- <br><div class="gmail_signature"><div dir="ltr"><div>Matthew Jordan<br></div><div>Digium, Inc. | Engineering Manager</div><div>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA</div><div>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></div></div></div>
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