google chrome - Asterisk sslv3 alert handshake failure -

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i using ubuntu v14.04.3 lts , asterisk 13.3.2. when try call extension sipml5 client play demo-congrats audio, call gets disconnected instantly. when check asterisk log, got following error:

[2016-08-24 06:07:49] error[31730][c-0000000c]: res_rtp_asterisk.c:2042 __rtp_recvfrom: dtls failure occurred on rtp instance '0x7f547c013c68' due reason 'sslv3 alert handshake failure', terminating [2016-08-24 06:07:49] warning[31730][c-0000000c]: res_rtp_asterisk.c:3911 ast_rtcp_read: rtcp read error: unspecified.  hanging up. [2016-08-24 06:07:49] warning[31730][c-0000000c]: app_playback.c:493 playback_exec: playback failed on sip/104600-00000007 /var/www/html/fetch_prompt [2016-08-24 06:07:49] error[31730][c-0000000c]: utils.c:1402 ast_carefulwrite: write() returned error: broken pipe

also using chrome v54.

i think error openssl, doesn't correct , complete answer yet solve issue. 1 know how solve issue?

solved issue upgrading openssl. use below commands upgrade openssl in ubuntu 14

# echo 'deb http://us.archive.ubuntu.com/ubuntu/ xenial main restricted universe multiverse' > /etc/apt/sources.list.d/xenial.list # aptitude update # aptitude install -y openssl libssl-dev # rm /etc/apt/sources.list.d/xenial.list # aptitude update

use below commands check openssl version

# ldd /usr/sbin/asterisk  | grep libssl libssl.so.1.0.0 => /lib/x86_64-linux-gnu/libssl.so.1.0.0 (0x00007f33ce117000)  # strings /lib/x86_64-linux-gnu/libssl.so.1.0.0 | grep 1.0.2 openssl_1.0.2 openssl_1.0.2g sslv3 part of openssl 1.0.2g-fips  1 mar 2016 tlsv1 part of openssl 1.0.2g-fips  1 mar 2016 dtlsv1 part of openssl 1.0.2g-fips  1 mar 2016 openssl 1.0.2g-fips  1 mar 2016  # openssl version openssl 1.0.2g-fips  1 mar 2016

after delete existing asterisk keys , recreate keys again

# rm /etc/asterisk/keys/* # cd /usr/src/astersik*/contrb/scripts # sudo ./ast_tls_cert -c pbx.mycompany.com -o "my super company" -d /etc/asterisk/keys # asterisk -rx "reload"

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