[Freeswitch-users] WebRTC calls failing

SamyGo govoiper at gmail.com
Fri Feb 10 03:05:11 MSK 2017


Hi,
I've trying to figure out the error that causes the WebRTC based calls to
hangup.

2017-02-09 18:57:53.708073 [NOTICE] switch_rtp.c:1275 Auto Changing audio
stun/rtp/dtls port from 70.54.102.180:56188 to 70.54.102.180:1572
2017-02-09 18:57:54.688079 [ERR] switch_rtp.c:3165 audio Handshake failure 1
2017-02-09 18:57:54.688079 [INFO] switch_rtp.c:3166 Changing audio DTLS
state from HANDSHAKE to FAIL

Operating System is: Ubuntu 14.04.5 LTS

*OpenSSL version:*
OpenSSL 1.0.1f 6 Jan 2014
built on: Mon Jan 30 20:38:38 UTC 2017
platform: debian-amd64
options:  bn(64,64) rc4(16x,int) des(idx,cisc,16,int) blowfish(idx)
compiler: cc -fPIC -DOPENSSL_PIC -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN
-DHAVE_DLFCN_H -m64 -DL_ENDIAN -DTERMIO -g -O2 -fstack-protector
--param=ssp-buffer-size=4 -Wformat -Werror=format-security
-D_FORTIFY_SOURCE=2 -Wl,-Bsymbolic-functions -Wl,-z,relro -Wa,--noexecstack
-Wall -DMD32_REG_T=int -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT
-DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM
-DSHA512_ASM -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM
-DGHASH_ASM
OPENSSLDIR: "/usr/lib/ssl"

Tried couple of Jira Bug links around this and still no progress. Disabling
"inbound_late_negotiations" doesn't help either.


Regard,
Sammy
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