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failure when RTP sequence number switches from 65535 -> 0 #29
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There isn't much detail in the asterisk bug report on this. The one comment about TLS and 3DES is a bit confusing since libsrtp supports neither of these. Perhaps the problem is in Asterisk. Given the limited information in the bug report, this looks like the roll-over counter (ROC) logic isn't working. If the ROC were broken in libsrtp, it's difficult to imagine that nobody else would have experienced the same problem. |
@jfigus this issue depens on the client implementation of srtp. |
Just as an FYI, my test suite using libsrtp has one test that does 1 million encode/decodes on the same stream, which obviously causes the sequence number to roll over around 16 times - This always succeeds... Implies the issue is not in libsrtp. Maybe the nokia phones are doing something nonstandard (but not forbidden by RTP) like changing the ssrc when the sequence number rolls over - I've seen similar oddities in the past, and that would certainly cause decryption to break. |
Alexander Traud (@traud) did some patches to fix this issue (see: https://issues.asterisk.org/jira/browse/ASTERISK-16898). |
I have tested the patch with Nokia E66. I had to modify the recognition string in the asterisk patch since the E66 uses "E66-1 RM-343 510.21.009". |
Closing this since there are no plans to patch libsrtp. |
See here https://issues.asterisk.org/jira/browse/ASTERISK-16898
This issue still exists and the claim is that it is caused by libsrtp. In my case I'm using the latest wheezy libsrtp 1.4.4+20100615~dfsg-2 on i386. As described in the link above there is a loss of audio on average around 10 min mark (depending on the initial sequence number) with Nokia phones <-> Asterisk that uses libsrtp. The audio is lost one way, the Nokia still receives and plays audio but Asterisk rejects incoming SRTP packets.
Can this be fixed in your release? Thanks.
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