<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">It does not enable ice by default. If it did, everyone would be having issues. What condition is it triggering to enable ice? Its possible the lines from ice to webrtc triggering each other are a bit crossed, but I would have to see specifics. Again. Please move this to jira so we can discuss and have history of what is found recorded.<div class=""><br class=""><div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Dec 17, 2014, at 9:55 AM, Michel Brabants <<a href="mailto:michel.brabants@gmail.com" class="">michel.brabants@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class=""><div class=""><div class="">Hello,<br class=""><br class="">I found the issue: 2 lines of code. I'll see if I can submit a patch, but I'm looking into it why it was added. The problem exists primarily because everywhere where a local sdp is generated, the function set_ice is called, while I don't want it (because I don't need it - nonat - and it generates, ice-not-ready-errors causing rtp to be dropped). The set_ice-function also sets the webrtc-flag (not sure why), causing dtls to become a requirement, which is not true in the current context. Anyway, this is nothing for this list, but I just want to add for any user currently encountering this problem.<br class=""></div>I'll do my best to generate a usefull patch.<br class=""><br class=""></div>Michel<br class=""></div><div class="gmail_extra"><br class=""><div class="gmail_quote">On Fri, Dec 12, 2014 at 7:06 PM, Michael Jerris <span dir="ltr" class=""><<a href="mailto:mike@jerris.com" target="_blank" class="">mike@jerris.com</a>></span> wrote:<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class=""><br class="">
> On Dec 12, 2014, at 10:23 AM, Michel Brabants <<a href="mailto:michel.brabants@gmail.com" class="">michel.brabants@gmail.com</a>> wrote:<br class="">
><br class="">
> Hello,<br class="">
><br class="">
> I recently started upgrading to FS 1.4, but I encountered 2 difficulties of which I'm still looking into one:<br class="">
><br class="">
> 1) DTLS-configuration seems to be required, although we don't use it currently. We use normal sip-profiles (no webrtc). The option to disable it, is "webrtc_enable_dtls=false", which can b set in the dialplan. But why is it trying to enable it by default? Can you disable it also in a profile?<br class="">
><br class="">
<br class="">
</span>In what way do you think that some configuration is required?<br class="">
<span class=""><br class="">
> 2) Also a change because of webrtc seemingly. When receiving an invite (without sdp - 3pcc-request), freeswitch in the end response in its 200 OK with a rtcp-mux-line in its sdp. We don't want rtcp-mux, just rtp-port+1 for rtcp. When looking at the code, I don't currently know why FS sends back the myx-parameter as it seems only enabled when the other ends proposes it or am I missing something?<br class="">
><br class="">
<br class="">
</span>Please report a bug on this.<br class="">
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