WebRTC using rtp_sdes_suites=AES_CM_128_HMAC_SHA1_80
Good Day, Running FS 1.6.17 on CentOS 7.4 We are running WebRTC and are required to use AEAD_AES_256_GCM_8 or AEAD_AES_128_GCM_8 for security. I have eliminated all of the options in the vars file except rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8. Calls are successfully completing, however, in debug we are seeing AES_CM_128_HMAC_SHA1_80 as the sdes suite for srtp:dtls. 2018-05-04 22:38:30.429310 [INFO] switch_rtp.c:3185 Changing audio DTLS state from HANDSHAKE to SETUP 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3094 audio Fingerprint Verified. 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3908 Activating audio Secure RTP SEND 2018-05-04 22:38:30.450549 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3886 Activating audio Secure RTP RECV 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3134 Changing audio DTLS state from SETUP to READY 2018-05-04 22:38:30.450549 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2018-05-04 22:38:30.450549 [DEBUG] switch_rtp.c:1885 rtcp_stats_init: audio ssrc[3910337773] base_seq[2433] Any ideas on how or where to change this to the desired encryption protocol? Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com<http://www.thenavisway.com/> Facebook<https://www.facebook.com/theNAVISway/> | Twitter<https://twitter.com/NAVISway> | LinkedIn<https://www.linkedin.com/company/navisway> | Blog<https://www.thenavisway.com/blog>
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Jerry Chinn