[asterisk-bugs] [JIRA] (ASTERISK-26207) [patch] sRTP: Count a roll-over of the sequence number even on lost packets.

[Stephan Eisvogel (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26207?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=231540#comment-231540 ] 

Stephan Eisvogel commented on ASTERISK-26207:
---------------------------------------------

Hey Alexander,

any chance this fix is causing huge false lost packet counts in sip show channelstats?

Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
217.0.23.100     p65562t1469  00:01:20 0000003774  0000000000 ( 0.00%) 0.0000 0000003775  0000001998 (52.93%) 0.0001
dropping after a while to this
217.0.23.100     p65562t1469  00:06:38 0000019663  0000000000 ( 0.00%) 0.0000 0000019663  0000001998 (10.16%) 0.0001

Should be zero because that is a Telekom upstream on unloaded VDSL.

Regards,
Stephan

> [patch] sRTP: Count a roll-over of the sequence number even on lost packets.
> ----------------------------------------------------------------------------
>
>                 Key: ASTERISK-26207
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26207
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 11.22.0, 13.9.1
>            Reporter: Alexander Traud
>            Assignee: Alexander Traud
>            Severity: Minor
>         Attachments: roc_0x7fff.patch
>
>
> See the sRTP FAQ, [Question 6…|http://srtp.sourceforge.net/faq.html#Q6]
> With RTP media, each packet gets its own sequence number to detect lost  and late packets. That sequence number is limited to a value of {{0xffff}}. To avoid any replay-attacks, sRTP introduced a counter which is incremented whenever the sequence number rolls over (ROC). The ROC is not transmitted. Therefore, both sender and receiver must note each roll over of the sequence number.
> The initial sequence number is created randomly. However, when
> - that random value is near the maximum value *and*
> - the first RTP packets get lost (or arrive late),
> the receiver might not notice the roll-over of the sequence number. The receiver might think, the RTP sequence started at 0,1,2,3…. Consequently, the ROC is of the receiver is 0. The ROC of the sender is 1. Both parties lost ROC synchronization. Therefore, the receiver is not able to decrypt the sRTP packets anymore. No media is the result.
> The mentioned webpage therefore suggests: The SRTP sender should randomly select an initial sequence number that is between {{0x0000}} and {{0x7fff}}. The attached patch does this.



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