From lmcdasi at yahoo.com  Wed Oct 11 19:33:39 2023
From: lmcdasi at yahoo.com (lmcdasi at yahoo.com)
Date: Wed, 11 Oct 2023 23:33:39 +0000 (UTC)
Subject: [libsrtp] Question about AUTH_FAIL
References: <1044684858.11211781.1697067219371.ref@mail.yahoo.com>
Message-ID: <1044684858.11211781.1697067219371@mail.yahoo.com>

Hi,
? ?I have a question regarding error code AUTH_FAIL during srtp_unprotect. I'm using gstreamer to send srtp audio packets and during transmission I do have a 'pause/resume functionality.?
? ?At pause the gst-launch cmd is terminated.
? ?At resume the gst-launch is restarted but by having a new SSRC (which is a mistake), while the srtp key remains the same and the sequence number is higher than the last one. I'm using:??AES_CM_128_HMAC_SHA1_32

? ?Since the SSRC has changed, should't the srtp library return an AUTH_FAIL ? It does not complain that the SSRC has changed and I do not understand why.
? ?Thank you.
BR,Dan S.
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From paulej at packetizer.com  Wed Oct 25 10:08:46 2023
From: paulej at packetizer.com (Paul E. Jones)
Date: Wed, 25 Oct 2023 14:08:46 +0000
Subject: [libsrtp] Question about AUTH_FAIL
In-Reply-To: <1044684858.11211781.1697067219371@mail.yahoo.com>
References: <1044684858.11211781.1697067219371.ref@mail.yahoo.com>
 <1044684858.11211781.1697067219371@mail.yahoo.com>
Message-ID: <embdc3c1d7-56dd-4187-8645-ca589b11c321@42d02c9f.com>

Dan,

When using ssrc_any_inbound or ssrc_any_outbound, I think this is 
expected.  With a new SSRC, I would assume the ROC is zero. So seeing a 
new SSRC within the context of the policy applied to a SRTP policy 
(srtp_policy_t), the library should automatically use the associated 
keys with that incoming SSRC.

Paul

------ Original Message ------
>From "lmcdasi--- via libsrtp" <libsrtp at lists.packetizer.com>
To "libsrtp at lists.packetizer.com" <libsrtp at lists.packetizer.com>
Date 10/11/2023 7:33:39 PM
Subject [libsrtp] Question about AUTH_FAIL

>Hi,
>
>    I have a question regarding error code AUTH_FAIL during 
>srtp_unprotect. I'm using gstreamer to send srtp audio packets and 
>during transmission I do have a 'pause/resume functionality.
>
>    At pause the gst-launch cmd is terminated.
>
>    At resume the gst-launch is restarted but by having a new SSRC 
>(which is a mistake), while the srtp key remains the same and the 
>sequence number is higher than the last one. I'm using:  
>AES_CM_128_HMAC_SHA1_32
>
>    Since the SSRC has changed, should't the srtp library return an 
>AUTH_FAIL ? It does not complain that the SSRC has changed and I do not 
>understand why.
>
>    Thank you.
>
>BR,
>Dan S.
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