Old openssl version maybe? <br><div class="gmail_quote"><div dir="ltr">Em qui, 4 de mai de 2017 às 11:19, sachin <<a href="mailto:snehsach@rediffmail.com">snehsach@rediffmail.com</a>> escreveu:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello All,<br><br>I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN.<br><br>The clients are getting registered over wss. I have created self signed certificates. In var.xml I have set the codecs setting as follows<br><br> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA"><br> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"><br><br>I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error<br><br>2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 1<br>2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL<br>2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001@df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]<br><br>My setup is as follows<br><br>SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone<br><br>I am attaching the logs for the reference.<br>fs-logleve9.txt : Debug trace with loglevel =9<br>fs-sip-trace.txt : Sip trace<br>Please let me know what could the issue and pointers to resolve the same.<br><br>Thanks and Regards<br>SD<br>_________________________________________________________________________<br>
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