[asterisk-bugs] [JIRA] Commented: (ASTERISK-20098) P2P bridging can cause the SSRC of a RTP session to change during a call

[David Woolley (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20098?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=194924#comment-194924 ] 

David Woolley commented on ASTERISK-20098:
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The only problems we have observered is when there is a discontinuity in the timestamps, but no corresponding change in the SSRC, specifically for an AgentLogin agent during the MoH to beep to through audio transition.  Without an SSRC change, Cisco phones on the far side of CUCM, but with direct media, buzz at the frame rate for some time.  I don't believe excess SSRC changes would cause problems.

> P2P bridging can cause the SSRC of a RTP session to change during a call
> ------------------------------------------------------------------------
>
>                 Key: ASTERISK-20098
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20098
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Core/RTP
>            Reporter: Morten Tryfoss
>            Severity: Minor
>         Attachments: p2pbridge.diff
>
>
> P2P bridging can cause the SSRC id for an RTP stream to change during an active call. This can make some equiptment unhappy.
> I've added a configuration option so that this can be turned off.
> See the attached patch.

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