[Freeswitch-users] audio Handshake failure 1

[Alex Polischuk]

Thanks


On Mon, Jul 27, 2015 at 5:45 PM, Javier Menendez <menendez.garcia at gmail.com>
wrote:

> Please check *https://freeswitch.org/jira/browse/FS-7839
> <https://freeswitch.org/jira/browse/FS-7839> *
>
> On Mon, Jul 27, 2015 at 8:33 AM, Alex Polischuk <alxpol at gmail.com> wrote:
>
>> Hi All,
>>
>> I use FreeSwitch for transcoding SRTP from WebRTC to SIP with RTP and
>> vice versa.
>> When Firefox version 38 add requires for Perfect Forward Secrecy (PFS) I
>> have problem with Handshake in FS when Firefox Browser terminate a call. I
>> found that that problem was fixed in the version 1.4.19
>> https://freeswitch.org/jira/browse/FS-7425
>>
>> I have the same problem in the version 1.4.20
>>
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
>> switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
>> switch_rtp.c:2832 audio Fingerprint Verified.
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
>> switch_rtp.c:3384 Activating Audio Secure RTP SEND
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
>> switch_rtp.c:3362 Activating Audio Secure RTP RECV
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
>> switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.604193 [DEBUG]
>> switch_rtp.c:1937 rtcp_stats_init: ssrc[-1374535478] base_seq[6179]
>> f01c46d8-33a0-11e5-ac83-891cb7dea826 2015-07-26 14:16:46.644210 [DEBUG]
>> switch_rtp.c:5884 Correct ip/port confirmed.
>> f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [ERR]
>> switch_rtp.c:2917 audio Handshake failure 1
>> f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [INFO]
>> switch_rtp.c:2918 Changing audio DTLS state from HANDSHAKE to FAIL
>> f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [NOTICE]
>> switch_rtp.c:2899 Hangup sofia/external/
>> 992555500004 at voip.webrtc.jajah.com [CS_EXCHANGE_MEDIA]
>> [DESTINATION_OUT_OF_ORDER]
>>
>>
>> what additional definition I should make to resolve this problem?
>>
>> Thanks,
>> Alex Polischuk
>>
>>
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>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
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>
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