Old openssl version maybe?

Em qui, 4 de mai de 2017 às 11:19, sachin <snehsach@rediffmail.com> escreveu:

Hello All,

​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.

The clients are getting registered over wss. I have created self signed certificates.  In var.xml I have set the codecs setting as follows

  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA">
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA">

​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

2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 1
2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL
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]

My setup is as follows

SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone

I am attaching the logs for the reference.
fs-logleve9.txt : Debug trace with loglevel =9
​fs-sip-trace.txt : Sip trace
Please let me know what could the issue and pointers to resolve the same.

​Thanks and Regards
SD

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