<p dir="ltr">Noel</p>
<p dir="ltr">That would be useful info. I also don't mind writing stuff up if I get to grips with it and its useful.</p>
<p dir="ltr">Initially though I need to understand why FreeSWITCH reports in its logs that the sdp is fine but it then zeroes out the port for the actual sdp it sends in the SIP</p>
<div class="gmail_quote">On 28 Jan 2014 14:55, "Noel Morgan" <<a href="mailto:noel@vwci.com">noel@vwci.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Pete,<div><br></div><div>Works great although some of the configuration is hard coded. You can get better performance changing things like complexity. </div><div><br></div><div>Did I just commit to writing a way to configure this? </div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jan 28, 2014 at 8:27 AM, Pete Kelly <span dir="ltr"><<a href="mailto:pkelly@gmail.com" target="_blank">pkelly@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">Hi<div><br></div><div>I am trying to use the opus codec built into FreeSWITCH to talk to the pjsip library (running on an Android handset).</div><div><br>There seems to be very little docs on opus support within FreeSWITCH outside the original post saying Anthony has got it working in 2011?</div>
<div><br></div><div>Does anyone have any experience with using this codec with FreeSWITCH?</div><div><br></div><div><br></div></div>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr"><p style="font-family:arial;font-size:small">Thanks much,</p><p style="font-family:arial;font-size:small">Noel Morgan<br>VoiceWARE Communications, Inc.<br>
Direct (214) 390-9113<br>Mobile (214) 789-7189<br>visit: <a href="http://www.vwci.com/" target="_blank">http://www.vwci.com/</a><br>email: <a href="mailto:noel@vwci.com" target="_blank">noel@vwci.com</a></p></div>
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