<div dir="ltr">Hey all,<div><br></div><div>Reviving this thread just to see if anyone has thoughts here?</div><div><br></div><div>I've tried digging through the source and can't seem to find where the re-opening of auto_adjust happens during a re-invite. As an aside bug (I'll report in JIRA) - it looks like the always_auto_adjust RTP bug results in very choppy audio.</div><div><br></div><div>That said, I don't need always_auto_adjust, as I'm fine with sending a re-invite when I need this to happen. It just doesn't seem to be working. Is there something else that needs to be set or is this also a bug?</div><div><br></div><div>Best,</div><div>Colin</div><br><div class="gmail_quote"><div dir="ltr">On Fri, May 13, 2016 at 9:34 PM Colin Morelli <<a href="mailto:colin.morelli@gmail.com" target="_blank">colin.morelli@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Is this the case even when the RTP IP in the SDP doesn't change? I'm seeing successful re-invites being processed with no auto adjust happening afterwards. I see:<div><br></div><div>2016-05-13 21:28:24.216684 [DEBUG] sofia.c:7614 Processing updated SDP<br></div><div><br></div><div>Indicating that FS did receive the SDP in the re-invite but nothing else about RTP auto adjust afterwards. This is on FreeSWITCH Version 1.6.7-14-d38d065~64bit (-14-d38d065 64bit) running on Debian 8 x86_64.</div><div><br></div><div>Thanks for the response.</div><div><br></div><div>Best,</div><div>Colin</div></div><br><div class="gmail_quote"><div dir="ltr">On Fri, May 13, 2016 at 9:14 PM Michael Jerris <<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">we already re-open the auto adjust window on reinvite <span></span><br><br>On Friday, May 13, 2016, Colin Morelli <<a href="mailto:colin.morelli@gmail.com" target="_blank">colin.morelli@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">I have mobile applications running behind NAT. When their reachability changes (and the device's local route updates), I want to automatically switch the RTP stream to the new address/port combination.<div><br></div><div>I've tried using the RTP ALWAYS_AUTO_ADJUST bug, but that results in <i>very</i> choppy audio. I also don't really need it, as the only time I care to perform RTP auto-adjust is after an invite session. For this case I can safely assume that the client <i>will</i> send a re-invite when its address changes. Is there any way to perform RTP audio adjust only on a re-invite?</div><div><br></div><div>Using STUN doesn't seem right for two reasons: 1) on its own it can't solve this problem (even if it determines the external IP, the port is still wrong). 2) it results in issues on FS "Invalid STUN/ICE packet received 20 bytes"</div><div><br></div><div>Would appreciate any help.</div><div><br></div><div>Best,</div><div>Colin</div></div>
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