WebRTC: "audio Handshake failure 1" on Chrome (59)

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Te Matau

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Jun 30, 2017, 12:40:04 AM6/30/17
to 2600hz-dev
When an outbound call is made using WebRTC the call can be answered successfully and then disconnected. However on a subsequent call the call disconnects immediately upon being answered.

This behaviour doesn't occur on Firefox (54). It does happen on Chrome (59).

Freeswitch log from a failed call:
1nvn9umrob00hdd5d68i 2017-06-29 23:52:22.159152 [ERR] switch_rtp.c:2917 audio Handshake failure 1
1nvn9umrob00hdd5d68i 2017-06-29 23:52:22.159152 [INFO] switch_rtp.c:2918 Changing audio DTLS state from HANDSHAKE to FAIL

Freeswitch log from a successful call:
1nvn96k703v5347bbmmn 2017-06-29 23:49:27.274108 [INFO] switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP
1nvn96k703v5347bbmmn 2017-06-29 23:49:27.274108 [INFO] switch_rtp.c:2832 audio Fingerprint Verified.
1nvn96k703v5347bbmmn 2017-06-29 23:49:27.274108 [INFO] switch_rtp.c:3394 Activating Audio Secure RTP SEND
1nvn96k703v5347bbmmn 2017-06-29 23:49:27.274108 [INFO] switch_rtp.c:3372 Activating Audio Secure RTP RECV
1nvn96k703v5347bbmmn 2017-06-29 23:49:27.274108 [INFO] switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY
1nvn96k703v5347bbmmn 2017-06-29 23:49:27.294127 [DEBUG] switch_rtp.c:1937 rtcp_stats_init: ssrc[-1624029046] base_seq[721]
1nvn96k703v5347bbmmn 2017-06-29 23:49:27.294127 [DEBUG] switch_core_io.c:528 Setting BUG Codec opus:116

Any suggestions on how to troubleshoot further appreciated.
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