<div dir="ltr"><div>Hi Babak,</div><div><br></div><div>thanks for the traces.</div><div><br></div><div>As you put it, problem is probably connected with some intricacies of dynamic mapping of codec in sdp. <br></div><div><br></div><div>I would do the stupid guy path: setup linphone (or some other well behaving client) to only use opus (or opus and g711, just to be webrtc compliant), and then make navamessenger do exactly like as linphone do.</div><div><br></div><div>-giovanni<br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, Apr 30, 2019 at 8:04 AM Babak Yakhchali <<a href="mailto:babak.freeswitch@gmail.com">babak.freeswitch@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div dir="ltr"><div dir="ltr">Hi again<div>here is the pastebin for originating from freeswitch to softphone:</div><div><a href="https://pastebin.freeswitch.org/view/3a084039" target="_blank">https://pastebin.freeswitch.org/view/3a084039</a><br></div><div>this is the one for incomming call from softphone:</div><div><a href="https://pastebin.freeswitch.org/view/6faf8a02" target="_blank">https://pastebin.freeswitch.org/view/6faf8a02</a><br></div></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Sat, Apr 27, 2019 at 8:23 PM Giovanni Maruzzelli <<a href="mailto:gmaruzz@gmail.com" target="_blank">gmaruzz@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div>please, can you post (or much better, pastebin) a SIP trace ?</div><div><br></div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Sat, Apr 27, 2019 at 1:54 PM Babak Yakhchali <<a href="mailto:babak.freeswitch@gmail.com" target="_blank">babak.freeswitch@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">Hi<div>I'm trying to make calls between a custom softphone as client and freeswitch using opus as codec. when client calls freeswitch everything is working fine. but when I originate a call from freeswitch there is no audio from sofphone but client hears fs. After capturing packets I see that when client sends invite it is using 107 as its payload type and freeswitch sends 107 in reply. But when fs sends invite it is using 102 and client is sending 107. What is the standard? I think the softphone is doing something wrong because testing with jitsi works fine.</div><div>thanks</div></div>
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<a href="https://freeswitch.com" rel="noreferrer" target="_blank">https://freeswitch.com</a></blockquote></div><br clear="all"><br>-- <br><div dir="ltr" class="gmail_signature">Sincerely,<br><br>Giovanni Maruzzelli<br>OpenTelecom.IT<br>cell: +39 347 266 56 18<br><br></div>