I still have the same problem.
Isn't there anyone who can help me?
Thanks
On Wed, 27 May 2020 at 10:48, Nuno Centeio <nuno.r.centeio@xxxxxxxxx> wrote:
Hi,I'm having a problem with PJSIP 2.9 in an Android.I've successfully built a SIP client but I'm having an issue when I receive a REINVITE with Replaces headers, where the RTP audio port switches from the initial invite.I've attached my log file.This is where I've received the REINVITE:10:07:43.167 pjsua_core.c .RX 1293 bytes Request msg INVITE/cseq=1 (rdata0x78a5706b30) from TCP X.X.X.X:5050:
INVITE sip:1012@X.X.X.X:5050;transport=TCP;ob SIP/2.0
From: "martinho2"<sip:1010@DomainXPTO>;tag=5bad1b70-b00000a-fd2-65014-2b863-7865b38a-2b863(...)Replaces: f3a2717b-4986-46e7-8a3c-7745f36a6284;to-tag=b003d99b-f717-4d0a-ae50-ad7977bda658;from-tag=5bad10b0-b00000a-fd2-65014-2b863-2cb0c32-2b863(...)m=audio 19768 RTP/AVP 111 103 104 9 0 8 106 105 13 110 112 113 126(...)In a server side Wireshark I see that my App is sending packets to the port 19714, which is a random port that was never negotiated.This doesn' seem to be a NAT problem as normal calls works perfectly.Can anyone point me in the right direction?Thanks
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