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.
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@lists.packetizer.com
To "libsrtp@lists.packetizer.com" libsrtp@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.
participants (2)
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lmcdasi@yahoo.com
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Paul E. Jones