please, can you post (or much better, pastebin) a SIP trace ? On Sat, Apr 27, 2019 at 1:54 PM Babak Yakhchali <babak.freeswitch@gmail.com> wrote:
Hi 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. thanks _________________________________________________________________________
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