Brian, isn't this solved in 1.6.16? - FS-10037 <http://t.sidekickopen68.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJN7t5XZsfmT10W5wLWvl4WYKY2W3MhGXn56dLgFf1Tdg-W02?t=https%3A%2F%2Ffreeswitch.org%2Fjira%2Fbrowse%2FFS-10037&si=5036397111279616&pi=a3abf16e-d54d-4338-ab43-13050b016003> [core] Update OpenSSL to version 1.0.2k for Windows 2017-05-05 15:26 GMT+02:00 Brian West <brian@freeswitch.org>:
OpenSSL on the windows build needs to be updated.
https://freeswitch.org/jira/browse/FS-9510
On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger <gregor@infomedia.si> wrote:
This error is familiar to me, I think so, if I remembered correctly.
I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it.
2017-05-04 19:15 GMT+02:00 Anthony Minessale <anthony.minessale@gmail.com
:
FS 1.5 sounds like a bad plan. Try latest FS 1.6 or master.
On Thu, May 4, 2017 at 12:04 PM, Ítalo Rossi <italo@freeswitch.org> wrote:
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.
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